Q1, You can not compile zaptel and asterisk
please make sure that:
1) You have installed all necessary packages and kernel source.
2) Make sure the version of kernel source is exactly same with the version of the kernel.
please check the few links:
http://wiki.openvox.cn/index.php/D110P
http://wiki.openvox.cn/index.php/D210P
http://wiki.openvox.cn/index.php/D410P
http://www.asteriskguru.com/tutorials/
3) make sure that you do not miss any packages or files in asterisk or zaptel.
4) make sure your system can access www.asterisk.org.
Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)
please check:
1) run lspci -vvvvv, make sure the system can detect the card. Tiger jet chip will be found. If there is no such Tiger jet chip, please clean the PCI slot and try again.
2) if lspc can find the card, make sure the pci id is included in the PCI table in our driver. how to patch the picid, please refer this link:
http://www.openvox.cn/kb/entry/2/
3) if step 1 and step 2 are ok, please check the zaptel.conf or system.conf to make sure that the setting is correct.
4) if step 3 is correct, please make sure that there is no mISDN tiger jet module in the system, if it is there, please remove that or add to blacklist.
5) if you still can not boot it up, you have to recompile zaptel or dahdi again.
Q3, You can not make calls from asterisk
there are few reasons why you can not make calls:
1) check your extensions from your asterisk side, make sure your sip is ready to make calls, and SIP is with a right context what you put in extensions.conf
2) your pri is and active(leds are in green).
3) leds are up and card driver has boot up properly, but the zapata.conf is
, so asterisk does not boot up properly,
please check by run: zap show channels
please check the pri status, it MUST be up and active
if is empty or no such command, you should check your zapata.conf
4) Make sure dmesg shows without any error
5) Make sure the pri is up and active without any error
6) Make sure the physical connection is well established
7) You maybe recompile your zaptel and asterisk again.
Q4, How can you set the digital card for your country?
To set the pbx with your country support, you must:
1) set timezone and defaultzone to your country in zaptel.conf or system.conf of dahdi
2) set the country=your country in indication.conf
Q5, How can you open the debug for asterisk?
1) You can edit the file logger.conf under /etc/asterisk,
enable the debug or error, those message will be stored under
/var/log/asterisk
2) you also can start your asterisk in this way:
asterisk -vvvvvvvvgc -d
Q6, How can i check the IRQ of analog cards?
please run the command:
cat /proc/interrupts
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.
more details, please check from here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Q7, Sound Quality Problems with Digital cards
please refer this link:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Q8, How can you compile asterisk with dahdi for D110P/D210P/D410P
please refer these links:
http://bbs.openvox.cn/viewthread.php?tid=576&extra=page%3D1
http://www.voip-info.org/wiki/view/DAHDI
http://www.russellbryant.net/blog/category/dahdi/
http://blog.paulsnet.org/?p=44
http://docs.tzafrir.org.il/dahdi-tools/?C=S%3BO=A
Q9, I am hearing an echo. What can I do to fix this?
please refer these links:
http://kb.digium.com/entry/1/
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
Q10, Asterisk does not properly detect when a caller hangs up the phone. How do I fix this?
please refer this link:
http://kb.digium.com/entry/6/
Q11, When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP?
For the TDM400P and TE110P cards, the LED's will not be lit up until the kernel module is loaded. The TDM400P LED's will light up when the ports are configured and the kernel module is loaded. They do not change if a phone or trunk is plugged in or not. The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is connected the LED should turn GREEN.
For the TE2XXP/TE4XXP the LED's should scroll(knightrider) RED even without the kernel module being loaded or anything plugged in. When you have the spans properly configured and kernel module loaded without a circuit or channel bank the LED's should pulse RED. With the module loaded and a circuit/channel bank connected they should be solid GREEN. link from here:
http://kb.digium.com/entry/13/
Q12, Why is my card getting an IRQ miss?
Each peice of hardware takes 1,000 interrupts per second. When, for some reason the cards get less than this, an IRQ miss occurs. You can see if the card is missing interrupts using 'zttool.'
IRQ misses can cause different problems with Asterisk. Symptoms of IRQ misses are bad audio quality or perhaps PRI errors, although IRQ misses will not cause alarms. Also DTMF detection not working is something that can be caused by IRQ misses as well.
Several common things that contribute to IRQ misses are: -Running the X window system -Shared IRQs -No hard drive DMA -Hard drive DMA too high (shoot for udma3) -Running serial terminals or frame buffers
To check for shared IRQs you can run:
1. cat /proc/interrupts
CPU0
0 10756672 XT-PIC timer 2 0 XT-PIC cascade 5 10812879 XT-PIC uhci_hcd, uhci_hcd, wctdm 10 226219 XT-PIC t1xxp, CS46XX 11 1550046 XT-PIC eth0, nvidia 12 387234 XT-PIC i8042 14 32641 XT-PIC ide0 15 18 XT-PIC ide1 NMI 0 LOC 10757616 ERR 40481 MIS 0
Notice the T100P card sharing with the sound card, and the TDM400P card is sharing with the USB controller. This will most likely cause problems. If you are not using any USB devices that would probably be ok, but it would be best to disable USB or get the card on it's own IRQ.
There are several ways to move cards to their own IRQ.
-Turn on APIC
-Tweak BIOS settings
-Try a different PCI slot
-Use setpci
refer this link from digium: http://kb.digium.com/entry/63/
good link:
http://wiki.telesoft.ro/mediawiki/index.php/AsteriskSoundProbemWithDigium.html
Q13, Why am I having DTMF detection problems?
Zaptel DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf. It may need to be turned on or off. If you need to force all DTMF detection to be done in software, you can set vpmdtmf support to 0 in wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime.
SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.
Q14, I am getting error messages about PCI Master Aborts. What is wrong?
This is a very rare case. When your computer's PCI subsystem experiences serious problems with OpenVox's cards upon initialization of the card, Linux will print out scrolling "PCI Master Abort" messages. What you should do is go into your system's BIOS, and turn off your motherboard's PNP (plug and play) feature. If this does not resolve your issue, You should contact OpenVox support.
Q15, list of asterisk pbx distributions
www.elastix.org
www.trixobx.org
http://www.briker.org/
http://www.easyasterisk.it/
http://pbxinaflash.org/
Q16, How can you install asterisk with Debian Ubutun
http://www.debianhelp.co.uk/asterisk.htm
http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu
http://ubuntuforums.org/showthread.php?t=136785
Q17, How can you install asterisk with Fedora?
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
http://www.asteriskguru.com/
Q18, How can you install asterisk with SuSe?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_suse.html
http://voip-manager.net/installation-linux-asterisk.php
Q19, install asterisk with Free BSD
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
http://www.voip-info.org/wiki/view/FreeBSD+zaptel
Q20, List of Asterisk OS Platforms
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms
Q21, Centos with asterisk
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos
Q22, digital cards with "TRUNK Dial failed due to CONGESTION" Problem
You must check:
1) your driver is loaded properly.
2) there is no error running dmesg with cards.
3) under asterisk console, run: zap show channels or dahdi show channels, make sure that there is no error
4) under asterisk console, run: pri show spans, make sure the spans are up and active
5) make sure your dialplan is set to a right channel.
Q23, How do you report a problem
In order to solve customer's problems very effective and efficiency,
when seeking a help from us, please give these information:
1) versions of kernel and Linux distribution
2) versions of asterisk and zaptel/dahdi
3) the name of cards used in your system
4) debug and error information from your system and asterisk
5) sending us zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf)
configuration files and extension.conf
6) after loading the driver, run the command: demsg and send the information to us
7) sending us the result of the command: cat /proc/interrupts
8) sending us the message of asterisk console when you making a call
9) inform the protocols you are using in your system
10) sending us a working ssh account with root permission if you need us to check the system.
11) making a backup for your important files
12) describing the problem in details
Q24,FATAL: Module wcte11xp not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the wcte11xp is under /lib/modules/2.6.XX/extra
Q25,FATAL: Module wct4xxp not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the wct4xxp.ko is under /lib/modules/2.6.XX/extra
Q26, Tools for PRI cards
you can use these tools to test the wctdm and opvxa1200
1) zttest
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
2) zttool
http://www.voip-info.org/wiki/view/Asterisk+zttool
3) ztmonitor
http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment
Q27,check information of wctdm.ko/wct4xxp.ko
Under /lib/modules/2.6.18-128.el5/misc
run command: modinfo wct4xxp.ko
Q28, How to debug wct4xxp
When loading the wct4xxp with a debug mode,
please loadding the driver in this way:
modprobe wct4xxp/wct1xxp debug=1 // open the debug and check the /var/log/message
Q29, RHEL/Centos 5.2: xpp/xdefs.h:117: error: conflicting types for ‘bool’
please refer this:
https://issues.asterisk.org/view.php?id=12889
Q30, xpp modules do not compile with kernel 2.6.19-1.2919.fc7
please refer this:
https://issues.asterisk.org/view.php?id=9006
Q31, spinlock.h error with RHEL 4
please refer this link:
http://forums.digium.com/viewtopic.php?p=17034&sid=c914a0a979f8437576c7aa92518fe48b
Q32, Compile error on CentOS-4.6 with Kernel-2.6.9-67.0.15.ELsmp and CONFIG_DAHDI_NET
please refer this link:
https://issues.asterisk.org/view.php?id=13427
Q33, dahdi_compat.h:31:27: error: zaptel/zaptel.h: No such file
please read this:
https://issues.asterisk.org/view.php?id=14121
Q34, when compiling zaptel, error: You do not appear to have the sources for...
please refer this:
http://forums.digium.com/viewtopic.php?t=7061
http://lists.digium.com/pipermail/asterisk-users/2007-June/189259.html
Q35, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol
please refer this:
http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2007-August/009225.html
Q36, How to install Octasic SoftEcho
please refer these links:
http://www.openvox.cn/download/user_manuals_english_version/Octvqeug_5000.pdf
http://www.octasic.com/en/products/softecho/softecho_asterisk.php
http://www.octasic.com/en/products/softecho/support.php
Q37, Bug in Zaptel 1.2.20.1 and 1.4.5.1 - Only MG2
please refer this:
http://trixbox.org/node/21080 http://www.rowetel.com/ucasterisk/oslec.html
Q38, Howto: OSLEC echo canceling + DAHDI 2.1.0.4 + Asterisk 1.4
please refer this:
http://www.asterisk.org/forum/viewtopic.php?p=125314&sid=9515c7b03cb14dc698e89467c3d49a86
Q39, Difference between zaptel and dahdi
please refer these links:
http://www.voip-info.org/wiki/view/DAHDI
http://docs.tzafrir.org.il/dahdi-linux/
http://docs.tzafrir.org.il/dahdi-tools/
Q40, Tonezones for zaptel.conf
The file zonedata.c contains the information about the tone zones used in libtonezone (and hence also in ztcfg). Here is a list of those zones:
us United States / North America
au Australia
fr France
nl Netherlands
uk United Kingdom
fi Finland
es Spain
jp Japan
no Norway
at Austria
nz New Zealand
it Italy
us-old United States Circa 1950 / North America
gr Greece
tw Taiwan
cl Chile
se Sweden
be Belgium
sg Singapore
il Israel
br Brazil
hu Hungary
lt Lithuania
pl Poland
za South Africa
pt Portugal
ee Estonia
mx Mexico
in India
de Germany
ch Switzerland
dk Denmark
cz Czech Republic
cn China
ar Argentina
my Malaysia
th Thailand
bg Bulgaria
ve Venezuela
ph Philippines
ru Russian Federation
tr Turkey
Q41, Tools from zaptel to dahdi
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool
zapconf -> dahdi_genconf (deprecates genzaptelconf)
Q42, Why are you unable to call out with Asterisk 1.4.22?
If you are using wctdm or opvxa1200 with Zaptel and Asterisk 1.4.22 then there is a known issue with outbound calls. The reason you are not able to call out is because Asterisk 1.4.22 has a new feature which detects if a analog line is plugged in or not, but this feature only works with Dahdi. So to fix the issue you can do one of the following.
edit the file under /asterisk-1.4.22、channels/chan_dahdi.c" find this line
1. ifdef DAHDI_CHECK_HOOKSTATE return 0;
2. else return 1;
Change the "0" to a "1"
1. ifdef DAHDI_CHECK_HOOKSTATE return 1;
2. else return 1;
Q43, Missing libpri
Symptom: chan_zap fails to load (no 'zap' in the CLI). In the logs you see the error:
chan_zap.c: Unknown signalling method 'pri_cpe'
Cause: chan_zap.so in Asterisk was built without support for libpri. libpri was not installed when you ran ./configure before building asterisk.
Fix: Rebuild asterisk and make sure libpri is supported.
$ strings channels/chan_zap.so | grep pri_cpe
pri_cpe
Q44, I have an E1/PRI line, incoming calls are working but outgoing calls are not working, what is wrong?
try to set:
pridialplan= local (or unknown, private, national, and international)
Q45, How to get more debug information
under asterisk console, run : pri intense debug span X, X is span number
Q46, T1/E1 Clock Synchronization
TE1 Clock synchronization is used to propagate a single clock source over the T1/E1 ports on a single card.
Before configuring your system you must identify which ports should be in NORMAL (slave) clock mode and which should be in MASTER clock mode.
All ports connected to TELCO MUST be in NORMAL mode,
because Telco is ALWAYS MASTER clock.
Example:
zaptel->Port 1 connected to TELCO // port 1 MUST be Normal(slave) clock mode
zaptel->Port 2 connected to channel bank or back to back to another T1/E1
device. In this scenario Port2 must be configured as CLOCK MASTER.
Q47, Cabling for PRI cards
the pin, 1,2,4,5 are used. ping 1 and 2 are used for rx; pin 4 and 5 are used for tx
please check from these links:
http://www.pbx.in/digium-te110p-loopback-cable-india-howto
http://www.chebucto.ns.ca/Chebucto/Technical/Manuals/Max/max6000/gs/cables.htm
http://help.pbxtra.com/Troubleshooting/How_to:_Perform_a_Pattern_Loopback_Test
http://www.voip-info.org/wiki/view/ztloop
http://help.pbxtra.com/Troubleshooting/T1_Cross-over_Cable
Q48, PRI cards working with MFC/R2
please refer these links:
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
http://zarzamora.com.mx/archivo-historico/48
Q49, PRI cards working with SS7
http://www.voip-info.org/wiki/view/Asterisk+SS7
http://www.cesnet.cz/doc/techzpravy/2007/asterisk-ss7-performance/asterisk-ss7-performance.pdf
http://www.openvox.cn/download/other_docs/ss71.pdf
http://www.pdf-search-engine.com/asterisk-ss7-pdf.html
http://www.openvox.cn/download/other_docs/Test%20chan_ss7.pdf
http://www.astricon.net/2008/glendale/web/presentations/>/Introduction_to_SS7_and_Asterisk_MFredrickson.pdf
中国ss7:http://bbs.openvox.cn/forumdisplay.php?fid=12
Q50, Resistance of PRI cards
the resistance of PRI cards from 75 OHM to 120 OHM.
Q51, BNC connector of PRI cards
If you requires a BNC connector, OpenVox will provides a BNC connector(Y cable)with PRI cards.
the longer cable is for TX
the shorter cable is for RX
Q52, chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
If you are being flooded (several times a second, non stop and the pri never worked) by lines as:
Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
Then probably the PRI you are using is not using PRI signalling but maybe some other type of signalling like E&M.
Q53, chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
If you see the error Jul 14 13:55:21 NOTICE[19519]:
chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
only occasionally, then you might have some devices in your pc (ide cards?) taking to long when taking an intterupt.
You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with
hdparm (make your drivers slower) or disable that card, and take a new one.
Q54, Signalling requested on channel 1 is E & M Wink but line is in PRI Signalling signalling.
If you see an error similar to this one:
Jul 14 14:06:58 ERROR[19635]: chan_zap.c:6593 mkintf:
Signalling requested on channel 1 is E & M Wink but line is in PRI Signalling signalling.
That means your zaptel.conf seems to be configured for PRI signalling (you defined B-channels and D-channels)
and in zapata.conf you didnt not put signalling=pri_cpe or pri_net, but em_w
Q55, Signalling requested on channel 24 is PRI Signalling but line is in Unknown signalling 896 signalling
Signalling requested on channel 24 is PRI Signalling but line is in Unknown
signalling 896 signalling
you defined in zapata.conf that channel 24 is also a b channel.
e.g.: channels = 1-24, while in zaptel.conf you had a line: dchannel=24
chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
only occasionally, then you might have some devices in your pc (ide cards?) taking to long when taking an intterupt.
You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with
hdparm (make your drivers slower) or disable that card, and take a new one.
Q56, chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
If you see the error Jul 14 13:55:21 NOTICE[19519]:
chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
only occasionally, then you might have some devices in your pc (ide cards?) taking to long when taking an intterupt.
You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with
hdparm (make your drivers slower) or disable that card, and take a new one.
Q57, How do I run a pattern loopback test (patlooptest) on my E1/T1 card?
OpenVox's pri cards can be tested using patlooptest. This test transmits a bit pattern and listens for the same bit pattern to be received, comparing the results. To run the test, plug an E1/T1 loopback cable into the port to be tested. here, the pin assignment of lookback cable is 1->4, 2->5. this is a example for how to make a lookback cable:
http://www.ortizonline.com/publications/april2004/loopback.htm
1. With the system powered off, confirm that the board jumpers are set appropriately for E1 mode or T1 mode. (Refer to the user's manual for the interface card.)
2. Use zttool (or dahdi_tool) to confirm that the span is "OK" (green alarm):
zttool
or
dahdi_tool
3. Make a backup of /etc/zaptel.conf (or /etc/dahdi/system.conf).
4. Configure the span(s) as follows.
For E1 mode, /etc/zaptel.conf (or /etc/dahdi/system.conf) should contain:
span=1,0,0,ccs,hdb3,crc4
clear=1-31
# for TE2xx and TE4xx, uncomment the following two lines
# span=2,0,0,ccs,hdb3,crc4
# clear=32-62
# for TE4xx, uncomment the following four lines
# span=3,0,0,ccs,hdb3,crc4
# clear=63-93
# span=4,0,0,ccs,hdb3,crc4
# clear=94-124
For T1 mode, /etc/zaptel.conf (or /etc/dahdi/system.conf) should contain:
span=1,0,0,esf,b8zs
clear=1-24
# for TE2xx and TE4xx, uncomment the following two lines
# span=2,0,0,esf,b8zs
# clear=25-48
# for TE4xx, uncomment the following four lines
# span=3,0,0,esf,b8zs
# clear=49-72
# span=4,0,0,esf,b8zs
# clear=73-96
5. Use ztcfg (or dahdi_cfg) to configure the channels:
ztcfg -vvv
or
dahdi_cfg -vvv
6. Make the tests, including patlooptest:
cd /usr/src/zaptel/
make tests
or
cd /usr/src/dahdi-linux-complete-N.N.N+N.N.N/tools/
make tests
7. Run patlooptest:
./patlooptest /dev/zap/1 300
Going for it...
or
./patlooptest /dev/dahdi/1 300
Going for it...
(To run patlooptest on each of four E1 spans:
./patlooptest /dev/zap/1 300
./patlooptest /dev/zap/32 300
./patlooptest /dev/zap/63 300
./patlooptest /dev/zap/94 300
or
./patlooptest /dev/dahdi/1 300
./patlooptest /dev/dahdi/32 300
./patlooptest /dev/dahdi/63 300
./patlooptest /dev/dahdi/94 300
To run patlooptest on each of four T1 spans:
./patlooptest /dev/zap/1 300
./patlooptest /dev/zap/25 300
./patlooptest /dev/zap/49 300
./patlooptest /dev/zap/73 300
or
./patlooptest /dev/dahdi/1 300
./patlooptest /dev/dahdi/25 300
./patlooptest /dev/dahdi/49 300
./patlooptest /dev/dahdi/73 300
)
The first parameter to patlooptest is the channel on which to run the test (e.g., /dev/zap/1 or /dev/dahdi/1 ). The test should be run on the first channel of the span in question.
The second parameter is an optional timeout (e.g., 300), measured in seconds.
patlooptest should complete without displaying any errors. If there are errors, it may indicate that the card or port is bad. However, errors could also be caused by interrupt misses or a faulty loopback plug.
If patlooptest does not terminate after the timeout interval, then the card is probably not taking any interrupts.
original link is from:http://kb.digium.com/entry/138/
Q58, How to install asterisk, zaptel and chan_ss7
if you want to install asterisk, chan_ss7 and zaptel, please follow these steps:
1) download asterisk-1.4.20, zaptel-1.4.10 and chan_ss7_1.1
2) unzip asterisk-1.4.20.tar.gz to /usr/src, under asterisk dir,
please run: ./configure, make , make install, make samples.
3) unzip zaptel-1.4.10.tar.gz to /usr/src/, under zaptel dir, please run:
./confiugre, make , make install
4) unzip the chan_ss7_1.1.tar.gz to /usr/src, under chan_ss7_1.1, please do this:
4.1) modify the Makefile, do like this:
# non-standard places.
INCLUDE+=-I /usr/src/zaptel-1.4.10/kernel ; point to the zaptel source
#INCLUDE+=-I../source/telephony/dahdi/include
INCLUDE+=-I /usr/src/asterisk-1.4.20 ; point to asterisk source
4.2) save and quit
4.3) run: make , make install ； 安装，编译 chan_ss7
4.4) copy ss7.conf file to /etc/asterisk/
4.5) copy chan_ss7.so to /usr/lib/asterisk/modules/
5) edit the zaptel.conf like this:
# Autogenerated by ./genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER)
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-31
# dchan=16
# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2"
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-62
# dchan=47
# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
span=3,3,0,ccs,hdb3,crc4
# termtype: te
bchan=63-93
# dchan=78
# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
span=4,4,0,ccs,hdb3,crc4
# termtype: te
bchan=94-124
#dchan=109
# Global data
loadzone = us
defaultzone = us
6) edit ss7.conf like this:
[linkset-ls1]
enabled => yes
enable_st => yes
use_connect => no
hunting_policy => even_mru
context => ss7_call
language => en
subservice => auto
variant => CHINA ; 支持中国ss7 号信令
[link-l1]
linkset => ls1
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes
echocancel => no
echocan_train => 350
echocan_taps => 128
[link-l2]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 33
enabled => yes
[link-l3]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 65
enabled => yes
[link-l4]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 97
enabled => yes
[host-openvox]； host IP
enabled => yes
opc => 0x222222 ; 点码
dpc => ls1:0x298922 ； 点码
links => l1:1
links => l1:1,l2:2,l3:3,l4:4
7) edit extensions.conf：
extensions.conf
[ss7_call]
exten => 100,1,Dial(ss7/outgoing number)
exten => 100,2,Hangup
8) Load zaptel, wct4xxp and start asterisk
modprobe zaptel ; load zaptel
modprobe wct4xxp ; load the wct4xxp
ztcfg -vvvvvvv ; load the channels
asterisk -vvvvvvvgc ; start asterisk
Under asterisk console
CLI> ss7 link status ；Check the status, make sure there is no error.
; 检查 ss7 的状态， 确保没有异常。
linkset ls1, link l1, schannel 16, sls 0, INSERVICE, rx: 5, tx: 1/3, sentseq/lastack: 4/4, total 199328, 199424
CLI> ss7 status
linkset idle busy initiating resetting total incoming total outgoing
ls1 30 0 0 0 0 0
gw1*CLI> ss7 linestat
Linkset: ls1
CIC 1 Idle
CIC 2 Idle
CIC 3 Idle
CIC 4 Idle
CIC 5 Idle
CIC 6 Idle
CIC 7 Idle
CIC 8 Idle
CIC 9 Idle
CIC 10 Idle
CIC 11 Idle
CIC 12 Idle
CIC 13 Idle
CIC 14 Idle
CIC 15 Idle
CIC 17 Idle
CIC 18 Idle
CIC 19 Idle
CIC 20 Idle
CIC 21 Idle
9） Use an extension dial 100 to make a call to ss7
Test tools
asterisk-1.4.20
zaptel-1.4.10
chan_ss7-1.1(1.1 以上版本已经支持中国ss7 号信令)， 1.1 以前的版本请到openvox.cn 下载。
Openvox D410P
References
http://www.dicea.dk/company/chan_ss7
voip-info.org
www.openvox.cn
Q59, How do I change the type of line from E1 to T1/J1 without using jumpers?
If you do not have physcial access to the card, and you need to set it in T1 or E1 mode you can pass a special option when loading the module to accomplish this task:
insmod wct4xxp t1e1override=0xFF
This will set all four spans into E1 mode.
insmod wct4xxp t1e1override=0x00
This will set all four spans into T1 mode.
If you need some spans T1 and some E1 use the following guidelines.
The settings of the spans must be passed as a bitmask in the module paramater. To create the bitmask use the table below adding the value of the numbers corrisponding to the spans you wish to enable. span value 1 1 2 2 3 4 4 8
So to enable spans 1 and 4 you would call modprobe like so:
modprobe wct4xxp t1e1override=0x09
As you can see I added 1 and 8 to get 9. Remember if you pass this value in HEX to format it properly.
To make spans 1, 2, and 4 E1 spans, you would use the following parameter:
... t1e1override=0x0B or ... t1e1override=11
url: http://kb.digium.com/entry/1/121/
Q60 To get the quick support, what information about the system I should send?
* Versions of kernel and Linux distribution
* Versions of asterisk and zaptel/dahdi
* Model of OpenVox products
* Debug and error information from your system and asterisk
* Zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf) configuration files and extension.conf if your have trouble with dialplan
* After loading the driver, run the command: demsg and send the information to us
* Output of the command: cat /proc/interrupts
* Message in asterisk console when you making a call, if you have trouble to drop/receive calls.
* Protocols you want to use(FXS,FXO,PRI..)
* Ssh account with root permission if you need us to check the system.
* Make a backup for your important files
* Describe the problem in details
* Serial numbers of OpenVox's products
Q61 How to install the openR2 with D115P on trixbox-2.8
1. Use command: yum install bison bison-devel ncurses ncurses-devel zlib zlib-devel openssl openssl-devel gnutls-devel gcc gcc-c++
2. copy this script to /usr/src/openvos.sh,then run the script to install the DE115P
cd /usr/src
wget http://downloads.openvox.cn/pub/drivers/dahdi-linux/openvox_dahdi-linux-2.2.0.tar.gz
tar -xvzf openvox_dahdi-linux-2.2.0.tar.gz
cd dahdi-linux-2.2.0
make
mkdir -p /lib/modules/`uname -r`/dahdi/opvxd115
cp /usr/src/dahdi-linux-2.2.0/drivers/dahdi/opvxd115/opvxd115.ko /lib/modules/`uname -r`/dahdi/opvxd115/
depmod -a
cd ..
wget http://downloads.openvox.cn/pub/firmwares/opvx-dahdi-fw-oct6114-032-1.07.01.tar.gz
tar -xzvf opvx-dahdi-fw-oct6114-032-1.07.01.tar.gz
cp dahdi-fw-oct6114-032.bin /lib/firmware/
mkdir -p /usr/lib/hotplug/firmware/
mv dahdi-fw-oct6114-032.bin /usr/lib/hotplug/firmware/
sed '/ztqoz\\/i\\t }elsif ($fqn =~ m{\\b(D115)\/.*}) {\n\t\t$type = "PRI";' /usr/lib/perl5/site_perl/5.8.8/Dahdi/Chans.pm > Chans.pm
mv Chans.pm /usr/lib/perl5/site_perl/5.8.8/Dahdi/
sed "/Wildcard TE122/i\ \t\t'D115P\/D115E \\\(PCI\/PCI-E\\\) Card'," /usr/lib/perl5/site_perl/5.8.8/Dahdi/Span.pm > Span.pm
mv Span.pm /usr/lib/perl5/site_perl/5.8.8/Dahdi/
sed "/d161:2400/i\ \t'1b74:0115' => { DRIVER => 'opvxd115', DESCRIPTION => 'OpenVox D115P/D115E '}," /usr/lib/perl5/site_perl/5.8.8/Dahdi/Hardware/PCI.pm > PCI.pm
mv PCI.pm /usr/lib/perl5/site_perl/5.8.8/Dahdi/Hardware/
echo "opvxd115" >> /etc/dahdi/modules
3. Download the openr2-1.3.tar.gz in /usr/src, use command:
./configure –prefix=/usr
make && make install
4. Download the asterisk-1.6.0.6 from www.asterisk.org
5. Download the the patch file from http://code.google.com/p/openr2/downloads/list?can=1&q=&colspec=Filename+Summary+Uploaded+Size
/usr/src/asterisk-1.6.0.6/openr2-asterisk-1.6.0.6.patch (download here)
then use command in /usr/src/asterisk-1.6.0.6:
patch -p0 < openr2-asterisk-1.6.0.6.patch
6. Download the autoconf-2.60 from http://www.gnu.org/software/autoconf/, extract autoconf-2.6.tar.gz, then compile it
./configure
make && make install
7. Use command: yum install automake
8. Use command in /usr/src/asterisk-1.6.0.6/:
./bootstrap.sh
./configure
make && make install
9. /etc/dahdi/system.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
dchan=16
cas=17-31:1101
loadzone = mx
defaultzone = mx
10. Please add this file in /etc/asterisk/chan_dahdi.conf
http://openr2.googlecode.com/files/chan_dahdi_or_zapata.conf.sample
11. In asterisk CLI use command: mfcr2 show channels
trixbox1*CLI> mfcr2 show channels
Chan Variant Max ANI Max DNIS ANI First Immediate Accept Tx CAS Rx CAS
1 MX 10 4 No No IDLE 0x00
2 MX 10 4 No No IDLE 0x00
3 MX 10 4 No No IDLE 0x00
4 MX 10 4 No No IDLE 0x00
5 MX 10 4 No No IDLE 0x00
6 MX 10 4 No No IDLE 0x00
7 MX 10 4 No No IDLE 0x00
8 MX 10 4 No No IDLE 0x00
9 MX 10 4 No No IDLE 0x00
10 MX 10 4 No No IDLE 0x00
11 MX 10 4 No No IDLE 0x00
12 MX 10 4 No No IDLE 0x00
13 MX 10 4 No No IDLE 0x00
14 MX 10 4 No No IDLE 0x00
15 MX 10 4 No No IDLE 0x00
17 MX 10 4 No No IDLE 0x00
18 MX 10 4 No No IDLE 0x00
19 MX 10 4 No No IDLE 0x00
20 MX 10 4 No No IDLE 0x00
21 MX 10 4 No No IDLE 0x00
22 MX 10 4 No No IDLE 0x00
23 MX 10 4 No No IDLE 0x00
24 MX 10 4 No No IDLE 0x00
25 MX 10 4 No No IDLE 0x00
26 MX 10 4 No No IDLE 0x00
27 MX 10 4 No No IDLE 0x00
28 MX 10 4 No No IDLE 0x00
29 MX 10 4 No No IDLE 0x00
30 MX 10 4 No No IDLE 0x00
31 MX 10 4 No No IDLE 0x00
Sunday, April 25, 2010
Troubleshooting of BRI cards
Q1, You can not compile zaptel/mISDN and asterisk
please make sure that:
1) You have installed all necessary packages and kernel source.
2) Make sure the version of kernel source is exactly same with the version of the kernel.
please check the few links:
http://www.openvox.cn/download/user_manuals_english_version/
http://www.asteriskguru.com/tutorials/
3) make sure that you do not miss any packages or files in asterisk or zaptel.
4) make sure your system can access www.asterisk.org.
Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)
if you are using bristuff with bri cards, please check:
1) run lspci -vvvvv, make sure the system can detect the card. Tiger jet chip will be found. If there is no such Tiger jet chip, please clean the PCI slot and try again.
2) if lspc can find the card, make sure the pci id is included in the PCI table in our driver. how to patch the picid, please refer this link:
http://www.openvox.cn/kb/entry/2/
3) if step 1 and step 2 are ok, please check the zaptel.conf or system.conf to make sure that the setting is correct.
4) if step 3 is correct, please make sure that there is no mISDN tiger jet module in the system, if it is there, please remove that or add to blacklist.
5) if you still can not boot it up, you have to recompile bristuff and asterisk again.
Q3, You can not make calls from asterisk
there are few reasons why you can not make calls:
1) check your extensions from your asterisk side, make sure your sip is ready to make calls, and SIP is with a right context what you put in extensions.conf
2) your qozap.ko or hfcmulti.ko does not boot up(leds are off).
3) leds are up and card driver has boot up properly, but the zapata.conf is
, so asterisk does not boot up properly,
please check by run: zap show channels
if is empty or no such command, you should check your zapata.conf/misdn.conf
4) if you are using mISDN, please make sure the misdn stack is up and active.
5) You maybe recompile your zaptel/mISDN and asterisk again.
Q4, How do you adjust the volume of voice for BRI cards?
You can edit rxgain and txgain in chan_misdn or zapata.conf.
Q5, list of drivers of BRI cards
Bri cards can work with these drivers:
1) mISDN-1_1_X， links:
http://wiki.openvox.cn/index.php/OpenVox_B200P_User_Manual_for_mISDN
http://www.openvox.cn/download/user_manuals_english_version
/B200P_B200E_B400P_B400E_User_Manual_mISDN.pdf
2) mISDN v2
http://www.openvox.cn/download/user_manuals_english_version
/B200%20B200E%20B400P%20B400E-User-Manual-mISDNv2.pdf
http://wiki.openvox.cn/index.php/OpenVox_B200P_User_Manual_for_mISDN(v2)
3) Bristuff
http://www.openvox.cn/download/user_manuals_english_version
/B200P_B400P_B400E_User_Manual_bristuff.pdf
http://www.openvox.cn/download/user_manuals_english_version
/B800P_User_Manual_bristuff.pdf
4) Dahdi wcb4xxp
http://bbs.openvox.cn/viewthread.php?tid=781&extra=page%3D1
5) isdn4bsd
http://forums.digium.com/viewtopic.php?t=21080
http://turbocat.net/~hselasky/isdn4bsd/
Before implementing one of these drivers, please make an investigation for your business requirement.
Q6, B200P_B400P_B800P pin assignment
if you want to check the pin assignment, please refer this:
http://bbs.openvox.cn/viewthread.php?tid=667&extra=page%3D2
Q7, Use PCM connection with OpenVox BRI cards.
if you want to use PCM cable to connect with the two BRI cards,
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=428&extra=page%3D3
Q8, netjetpci causes a problem with mISDN(B400P)+Zaptel(A400P)
If you conbine the bri card and analog card, please remove the netjetpci,
this is a solution to solve the problem:
http://bbs.openvox.cn/viewthread.php?tid=407&extra=page%3D3
Q9, How can you open the debug for mISDN/asterisk?
1) You can edit the file logger.conf under /etc/asterisk,
enable the debug or error, those message will be stored under
/var/log/asterisk
2) you also can start your asterisk in this way:
asterisk -vvvvvvvvgc -d
3) if you want to debug the mISDN, please enable the debug in mISDN.conf
Q10, How can i check the IRQ of BRI cards?
please run the command:
cat /proc/interrupts
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.
more details, please check from here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Q11, the definations of NT and TE
the bri cards support NT and TE mode. you can refer these links for the definitions:
http://www.techfest.com/networking/wan/isdn.htm
http://www.cisco.com/en/US/tech/tk652/tk653
/technologies_tech_note09186a0080111b16.shtml
http://www.asteriskguru.com/tutorials/bri.html
Q12, Sound Quality Problems with Analog cards
please refer this link:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Q13, MISDN with asterisk/trixbox/elastix
please refer these links:
http://misdn.org/index.php/MISDN_with_Asterisk
http://misdn.org/index.php/FAQ_chan_mISDN
http://www.voip-info.org/wiki/view/chan_misdn
http://trixbox.org/wiki/chan-misdn
http://www.openvox.cn/download/other_docs
/Integration%20B100P%20B200P%20B400P%20B800P%20with%20mISDN.pdf
http://www.servercare.nl/Lists/Posts/Post.aspx?ID=34
http://www.freepbx.org/forum/freepbx/users/elastix-and-idsn
Q14, I am hearing an echo. What can I do to fix this?
please refer these links:
http://kb.digium.com/entry/1/
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
Q15, zaphfc: dropped audio (z1=2331, z2=2314, wanted 8 got 17, dropped 9)
you can have a try to set please add apm=off in your grub.conf
Q16, When will the LED's light up on my BRI cards
Due to the difference of drivers, the LEDs of cards are different.
1) If the drivers are loaded correctly, the LEDs are in RED color.
2) If the layer1 and layer2 are up, and a conversation is established, the leds will be in GREEN.
3) B100P does not support LEDs
Q17, Why is my card getting an IRQ miss?
Each peice of hardware takes 1,000 interrupts per second. When, for some reason the cards get less than this, an IRQ miss occurs. You can see if the card is missing interrupts using 'zttool.'
IRQ misses can cause different problems with Asterisk. Symptoms of IRQ misses are bad audio quality or perhaps PRI errors, although IRQ misses will not cause alarms. Also DTMF detection not working is something that can be caused by IRQ misses as well.
Several common things that contribute to IRQ misses are: -Running the X window system -Shared IRQs -No hard drive DMA -Hard drive DMA too high (shoot for udma3) -Running serial terminals or frame buffers
To check for shared IRQs you can run:
1. cat /proc/interrupts
CPU0
0 10756672 XT-PIC timer 2 0 XT-PIC cascade 5 10812879 XT-PIC uhci_hcd, uhci_hcd, wctdm 10 226219 XT-PIC t1xxp, CS46XX 11 1550046 XT-PIC eth0, nvidia 12 387234 XT-PIC i8042 14 32641 XT-PIC ide0 15 18 XT-PIC ide1 NMI 0 LOC 10757616 ERR 40481 MIS 0
Notice the T100P card sharing with the sound card, and the TDM400P card is sharing with the USB controller. This will most likely cause problems. If you are not using any USB devices that would probably be ok, but it would be best to disable USB or get the card on it's own IRQ.
There are several ways to move cards to their own IRQ.
-Turn on APIC
-Tweak BIOS settings
-Try a different PCI slot
-Use setpci
refer this link from digium: http://kb.digium.com/entry/63/
Q18, Why am I having DTMF detection problems?
Zaptel DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf. It may need to be turned on or off. If you need to force all DTMF detection to be done in software, you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime.
SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.
Q19, I am getting error messages about PCI Master Aborts. What is wrong?
This is a very rare case. When your computer's PCI subsystem experiences serious problems with OpenVox's cards upon initialization of the card, Linux will print out scrolling "PCI Master Abort" messages. What you should do is go into your system's BIOS, and turn off your motherboard's PNP (plug and play) feature. If this does not resolve your issue, You should contact OpenVox support.
Q20, issues with wcb4xxp
this is a link with dahdi wcb4xxp, you many refer these links:
https://issues.asterisk.org/view.php?id=14834
https://issues.asterisk.org/view.php?id=14031
http://docs.tzafrir.org.il/dahdi-linux/README.html
https://bugs.digium.com/view.php?id=14031
https://bugs.digium.com/view.php?id=13897
Q21, list of asterisk pbx distributions:
www.elastix.org
www.trixobx.org
Q22, How can you install asterisk with Debian Ubutun
http://www.debianhelp.co.uk/asterisk.htm
http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu
http://ubuntuforums.org/showthread.php?t=136785
http://ww2.eq.uc.pt/servicos/laca/recursos/debian/voip-resources/installation-of-misdn
Q23, How can you install asterisk with Fedora?
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
http://www.asteriskguru.com/
Q24, How can you install asterisk with SuSe?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_suse.html
http://voip-manager.net/installation-linux-asterisk.php
Q25, install asterisk with Free BSD
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
http://www.voip-info.org/wiki/view/FreeBSD+zaptel
Q26, List of Asterisk OS Platforms
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms
Q27, Centos with asterisk
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos
http://www.thanosk.net/node/7
Q28, How do you report a problem
In order to solve customer's problems very effective and efficiency,
when seeking a help from us, please give these information:
1) versions of kernel and Linux distribution
2) versions of asterisk and zaptel/dahdi
3) the name of cards used in your system
4) debug and error information from your system and asterisk
5) sending us zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf)
configuration files and extension.conf
6) after loading the driver, run the command: demsg and send the information to us
7) sending us the result of the command: cat /proc/interrupts
8) sending us the message of asterisk console when you making a call
9) inform the protocols you are using in your system
10) send us a working ssh account with root permission if you need us to check the system.
11) make a backup for your important files
12) describe the problem in details
13) if you are using mISDN, please open the debug mode and run dmesg to get the
error or log message.
Q29,FATAL: Module qozap.ko cannot be found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the qozap.ko is under /lib/modules/2.6.XX/extra
Q30,FATAL: Module zaphfc.ko not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the zaphfc.ko is under /lib/modules/2.6.XX/extra
Q31, Disable Hisax and isdn in Centos/debian
If you are running a BRI card in the centos/FC or debian, please remove
Hisax and ISDN service from your system, add those services or modules into the system black list.
those packages will be conflict with mISDN/Bristuff/wcb4xxpdriver.
more details, please check the link:
http://pbxinaflash.com/forum/showthread.php?t=1418
Q32, How to debug brisutff/mISDN
When loading the qozap.ko/zaphfc.ko with a debug mode,
please loadding the driver in this way:
modprobe qozap,ko debug=1 // open the debug and check the /var/log/message
if you run misdn, please enable the debug in misdn.conf
Q33, RHEL/Centos 5.2: xpp/xdefs.h:117: error: conflicting types for ‘bool’
please refer this:
https://issues.asterisk.org/view.php?id=12889
Q34, xpp modules do not compile with kernel 2.6.19-1.2919.fc7
please refer this:
https://issues.asterisk.org/view.php?id=9006
Q35, spinlock.h error with RHEL 4
please refer this link:
http://forums.digium.com/viewtopic.php?p=17034&sid=c914a0a979f8437576c7aa92518fe48b
Q36, Compile error on CentOS-4.6 with Kernel-2.6.9-67.0.15.ELsmp and CONFIG_DAHDI_NET
please refer this link:
https://issues.asterisk.org/view.php?id=13427
Q37, dahdi_compat.h:31:27: error: zaptel/zaptel.h: No such file
please read this:
https://issues.asterisk.org/view.php?id=14121
Q38, when compiling zaptel/bristuff, error: You do not appear to have the sources for...
please refer these links:
http://forums.digium.com/viewtopic.php?t=7061
http://lists.digium.com/pipermail/asterisk-users/2007-June/189259.html
Q39, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol
please refer this:
http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2007-August/009225.html
Q40, How to install Octasic SoftEcho
please refer these links:
http://www.openvox.cn/download/user_manuals_english_version/Octvqeug_5000.pdf
http://www.octasic.com/en/products/softecho/softecho_asterisk.php
http://www.octasic.com/en/products/softecho/support.php
Q41, Bug in Zaptel 1.2.20.1 and 1.4.5.1 - Only MG2
please refer this:
http://trixbox.org/node/21080 http://www.rowetel.com/ucasterisk/oslec.html
Q42, Howto: OSLEC echo canceling + DAHDI 2.1.0.4 + Asterisk 1.4
please refer this:
http://www.asterisk.org/forum/viewtopic.php?p=125314&sid=9515c7b03cb14dc698e89467c3d49a86
Q43, Difference between zaptel and dahdi
please refer these links:
http://www.voip-info.org/wiki/view/DAHDI
http://docs.tzafrir.org.il/dahdi-linux/
http://docs.tzafrir.org.il/dahdi-tools/
Q44, Tonezones for zaptel.conf
The file zonedata.c contains the information about the tone zones used in libtonezone (and hence also in ztcfg). Here is a list of those zones:
us United States / North America
au Australia
fr France
nl Netherlands
uk United Kingdom
fi Finland
es Spain
jp Japan
no Norway
at Austria
nz New Zealand
it Italy
us-old United States Circa 1950 / North America
gr Greece
tw Taiwan
cl Chile
se Sweden
be Belgium
sg Singapore
il Israel
br Brazil
hu Hungary
lt Lithuania
pl Poland
za South Africa
pt Portugal
ee Estonia
mx Mexico
in India
de Germany
ch Switzerland
dk Denmark
cz Czech Republic
cn China
ar Argentina
my Malaysia
th Thailand
bg Bulgaria
ve Venezuela
ph Philippines
ru Russian Federation
tr Turkey
Q45, Tools from zaptel to dahdi
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool
zapconf -> dahdi_genconf (deprecates genzaptelconf)
Q46, Why are you unable to call out with Asterisk 1.4.22?
If you are using wctdm or opvxa1200 with Zaptel and Asterisk 1.4.22 then there is a known issue with outbound calls. The reason you are not able to call out is because Asterisk 1.4.22 has a new feature which detects if a analog line is plugged in or not, but this feature only works with Dahdi. So to fix the issue you can do one of the following.
edit the file under /asterisk-1.4.22、channels/chan_dahdi.c" find this line
1. ifdef DAHDI_CHECK_HOOKSTATE return 0;
2. else return 1;
Change the "0" to a "1"
1. ifdef DAHDI_CHECK_HOOKSTATE return 1;
2. else return 1;
Q47, install Asterisk zaphfc
please refer these links:
http://www.voip-info.org/wiki/view/Asterisk+zaphfc+install
http://www.openvox.cn/download/user_manuals_english_version/B100P_User_Manual_bristuff.pdf
Q48, Install bri card and analog cards in one system
if you want to install analog cards and bri cards in one system, it can be done in two ways:
1) install bristuff for both of analog cards and bri cards.
2) install mISDN with bri cards and zaptel with analog cards
please refer this link:
http://www.howtoforge.com/asterisk-zaptel-libpri-misdn-asterisk-addons-asterisk-gui-on-debian-etch
Q49, mISDN v2(LCR) with debian/asterisk
please refer the link:
http://blog.runtux.com/2009/03/09/61/
Q50, patch---mISDN rejects incoming calls
misdn rejects incoming calls, please refer this patch:
https://issues.asterisk.org/view.php?id=13488
Q51, B200P/B400P/B100P with trixbox 2.8 can't load chan_misdn.so
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=890&extra=page%3D1
Q52, How to install B200P/B400P/B100P/B800P with trixbox 2.6
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=1016&page=1&extra=page%3D1
Q53, How to install Ubuntu Server 9.04 with Asterisk and OpenVox B800P
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=1025&page=1&extra=page%3D1#pid4369
Q54, what kinds of protocols does mISDN support
mISDN so far, supports DSS1(EURO ISDN), it does not support NI1(National ISDN 1).
Q55, To get the quick support, what information about the system I should send?
* Versions of kernel and Linux distribution
* Versions of asterisk and zaptel/dahdi
* Model of OpenVox products
* Debug and error information from your system and asterisk
* Zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf) configuration files and extension.conf if your have trouble with dialplan
* After loading the driver, run the command: demsg and send the information to us
* Output of the command: cat /proc/interrupts
* Message in asterisk console when you making a call, if you have trouble to drop/receive calls.
* Protocols you want to use(FXS,FXO,PRI..)
* Ssh account with root permission if you need us to check the system.
* Make a backup for your important files
* Describe the problem in details
* Serial numbers of OpenVox's products
please make sure that:
1) You have installed all necessary packages and kernel source.
2) Make sure the version of kernel source is exactly same with the version of the kernel.
please check the few links:
http://www.openvox.cn/download/user_manuals_english_version/
http://www.asteriskguru.com/tutorials/
3) make sure that you do not miss any packages or files in asterisk or zaptel.
4) make sure your system can access www.asterisk.org.
Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)
if you are using bristuff with bri cards, please check:
1) run lspci -vvvvv, make sure the system can detect the card. Tiger jet chip will be found. If there is no such Tiger jet chip, please clean the PCI slot and try again.
2) if lspc can find the card, make sure the pci id is included in the PCI table in our driver. how to patch the picid, please refer this link:
http://www.openvox.cn/kb/entry/2/
3) if step 1 and step 2 are ok, please check the zaptel.conf or system.conf to make sure that the setting is correct.
4) if step 3 is correct, please make sure that there is no mISDN tiger jet module in the system, if it is there, please remove that or add to blacklist.
5) if you still can not boot it up, you have to recompile bristuff and asterisk again.
Q3, You can not make calls from asterisk
there are few reasons why you can not make calls:
1) check your extensions from your asterisk side, make sure your sip is ready to make calls, and SIP is with a right context what you put in extensions.conf
2) your qozap.ko or hfcmulti.ko does not boot up(leds are off).
3) leds are up and card driver has boot up properly, but the zapata.conf is
, so asterisk does not boot up properly,
please check by run: zap show channels
if is empty or no such command, you should check your zapata.conf/misdn.conf
4) if you are using mISDN, please make sure the misdn stack is up and active.
5) You maybe recompile your zaptel/mISDN and asterisk again.
Q4, How do you adjust the volume of voice for BRI cards?
You can edit rxgain and txgain in chan_misdn or zapata.conf.
Q5, list of drivers of BRI cards
Bri cards can work with these drivers:
1) mISDN-1_1_X， links:
http://wiki.openvox.cn/index.php/OpenVox_B200P_User_Manual_for_mISDN
http://www.openvox.cn/download/user_manuals_english_version
/B200P_B200E_B400P_B400E_User_Manual_mISDN.pdf
2) mISDN v2
http://www.openvox.cn/download/user_manuals_english_version
/B200%20B200E%20B400P%20B400E-User-Manual-mISDNv2.pdf
http://wiki.openvox.cn/index.php/OpenVox_B200P_User_Manual_for_mISDN(v2)
3) Bristuff
http://www.openvox.cn/download/user_manuals_english_version
/B200P_B400P_B400E_User_Manual_bristuff.pdf
http://www.openvox.cn/download/user_manuals_english_version
/B800P_User_Manual_bristuff.pdf
4) Dahdi wcb4xxp
http://bbs.openvox.cn/viewthread.php?tid=781&extra=page%3D1
5) isdn4bsd
http://forums.digium.com/viewtopic.php?t=21080
http://turbocat.net/~hselasky/isdn4bsd/
Before implementing one of these drivers, please make an investigation for your business requirement.
Q6, B200P_B400P_B800P pin assignment
if you want to check the pin assignment, please refer this:
http://bbs.openvox.cn/viewthread.php?tid=667&extra=page%3D2
Q7, Use PCM connection with OpenVox BRI cards.
if you want to use PCM cable to connect with the two BRI cards,
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=428&extra=page%3D3
Q8, netjetpci causes a problem with mISDN(B400P)+Zaptel(A400P)
If you conbine the bri card and analog card, please remove the netjetpci,
this is a solution to solve the problem:
http://bbs.openvox.cn/viewthread.php?tid=407&extra=page%3D3
Q9, How can you open the debug for mISDN/asterisk?
1) You can edit the file logger.conf under /etc/asterisk,
enable the debug or error, those message will be stored under
/var/log/asterisk
2) you also can start your asterisk in this way:
asterisk -vvvvvvvvgc -d
3) if you want to debug the mISDN, please enable the debug in mISDN.conf
Q10, How can i check the IRQ of BRI cards?
please run the command:
cat /proc/interrupts
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.
more details, please check from here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Q11, the definations of NT and TE
the bri cards support NT and TE mode. you can refer these links for the definitions:
http://www.techfest.com/networking/wan/isdn.htm
http://www.cisco.com/en/US/tech/tk652/tk653
/technologies_tech_note09186a0080111b16.shtml
http://www.asteriskguru.com/tutorials/bri.html
Q12, Sound Quality Problems with Analog cards
please refer this link:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Q13, MISDN with asterisk/trixbox/elastix
please refer these links:
http://misdn.org/index.php/MISDN_with_Asterisk
http://misdn.org/index.php/FAQ_chan_mISDN
http://www.voip-info.org/wiki/view/chan_misdn
http://trixbox.org/wiki/chan-misdn
http://www.openvox.cn/download/other_docs
/Integration%20B100P%20B200P%20B400P%20B800P%20with%20mISDN.pdf
http://www.servercare.nl/Lists/Posts/Post.aspx?ID=34
http://www.freepbx.org/forum/freepbx/users/elastix-and-idsn
Q14, I am hearing an echo. What can I do to fix this?
please refer these links:
http://kb.digium.com/entry/1/
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
Q15, zaphfc: dropped audio (z1=2331, z2=2314, wanted 8 got 17, dropped 9)
you can have a try to set please add apm=off in your grub.conf
Q16, When will the LED's light up on my BRI cards
Due to the difference of drivers, the LEDs of cards are different.
1) If the drivers are loaded correctly, the LEDs are in RED color.
2) If the layer1 and layer2 are up, and a conversation is established, the leds will be in GREEN.
3) B100P does not support LEDs
Q17, Why is my card getting an IRQ miss?
Each peice of hardware takes 1,000 interrupts per second. When, for some reason the cards get less than this, an IRQ miss occurs. You can see if the card is missing interrupts using 'zttool.'
IRQ misses can cause different problems with Asterisk. Symptoms of IRQ misses are bad audio quality or perhaps PRI errors, although IRQ misses will not cause alarms. Also DTMF detection not working is something that can be caused by IRQ misses as well.
Several common things that contribute to IRQ misses are: -Running the X window system -Shared IRQs -No hard drive DMA -Hard drive DMA too high (shoot for udma3) -Running serial terminals or frame buffers
To check for shared IRQs you can run:
1. cat /proc/interrupts
CPU0
0 10756672 XT-PIC timer 2 0 XT-PIC cascade 5 10812879 XT-PIC uhci_hcd, uhci_hcd, wctdm 10 226219 XT-PIC t1xxp, CS46XX 11 1550046 XT-PIC eth0, nvidia 12 387234 XT-PIC i8042 14 32641 XT-PIC ide0 15 18 XT-PIC ide1 NMI 0 LOC 10757616 ERR 40481 MIS 0
Notice the T100P card sharing with the sound card, and the TDM400P card is sharing with the USB controller. This will most likely cause problems. If you are not using any USB devices that would probably be ok, but it would be best to disable USB or get the card on it's own IRQ.
There are several ways to move cards to their own IRQ.
-Turn on APIC
-Tweak BIOS settings
-Try a different PCI slot
-Use setpci
refer this link from digium: http://kb.digium.com/entry/63/
Q18, Why am I having DTMF detection problems?
Zaptel DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf. It may need to be turned on or off. If you need to force all DTMF detection to be done in software, you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime.
SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.
Q19, I am getting error messages about PCI Master Aborts. What is wrong?
This is a very rare case. When your computer's PCI subsystem experiences serious problems with OpenVox's cards upon initialization of the card, Linux will print out scrolling "PCI Master Abort" messages. What you should do is go into your system's BIOS, and turn off your motherboard's PNP (plug and play) feature. If this does not resolve your issue, You should contact OpenVox support.
Q20, issues with wcb4xxp
this is a link with dahdi wcb4xxp, you many refer these links:
https://issues.asterisk.org/view.php?id=14834
https://issues.asterisk.org/view.php?id=14031
http://docs.tzafrir.org.il/dahdi-linux/README.html
https://bugs.digium.com/view.php?id=14031
https://bugs.digium.com/view.php?id=13897
Q21, list of asterisk pbx distributions:
www.elastix.org
www.trixobx.org
Q22, How can you install asterisk with Debian Ubutun
http://www.debianhelp.co.uk/asterisk.htm
http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu
http://ubuntuforums.org/showthread.php?t=136785
http://ww2.eq.uc.pt/servicos/laca/recursos/debian/voip-resources/installation-of-misdn
Q23, How can you install asterisk with Fedora?
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
http://www.asteriskguru.com/
Q24, How can you install asterisk with SuSe?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_suse.html
http://voip-manager.net/installation-linux-asterisk.php
Q25, install asterisk with Free BSD
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
http://www.voip-info.org/wiki/view/FreeBSD+zaptel
Q26, List of Asterisk OS Platforms
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms
Q27, Centos with asterisk
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos
http://www.thanosk.net/node/7
Q28, How do you report a problem
In order to solve customer's problems very effective and efficiency,
when seeking a help from us, please give these information:
1) versions of kernel and Linux distribution
2) versions of asterisk and zaptel/dahdi
3) the name of cards used in your system
4) debug and error information from your system and asterisk
5) sending us zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf)
configuration files and extension.conf
6) after loading the driver, run the command: demsg and send the information to us
7) sending us the result of the command: cat /proc/interrupts
8) sending us the message of asterisk console when you making a call
9) inform the protocols you are using in your system
10) send us a working ssh account with root permission if you need us to check the system.
11) make a backup for your important files
12) describe the problem in details
13) if you are using mISDN, please open the debug mode and run dmesg to get the
error or log message.
Q29,FATAL: Module qozap.ko cannot be found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the qozap.ko is under /lib/modules/2.6.XX/extra
Q30,FATAL: Module zaphfc.ko not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the zaphfc.ko is under /lib/modules/2.6.XX/extra
Q31, Disable Hisax and isdn in Centos/debian
If you are running a BRI card in the centos/FC or debian, please remove
Hisax and ISDN service from your system, add those services or modules into the system black list.
those packages will be conflict with mISDN/Bristuff/wcb4xxpdriver.
more details, please check the link:
http://pbxinaflash.com/forum/showthread.php?t=1418
Q32, How to debug brisutff/mISDN
When loading the qozap.ko/zaphfc.ko with a debug mode,
please loadding the driver in this way:
modprobe qozap,ko debug=1 // open the debug and check the /var/log/message
if you run misdn, please enable the debug in misdn.conf
Q33, RHEL/Centos 5.2: xpp/xdefs.h:117: error: conflicting types for ‘bool’
please refer this:
https://issues.asterisk.org/view.php?id=12889
Q34, xpp modules do not compile with kernel 2.6.19-1.2919.fc7
please refer this:
https://issues.asterisk.org/view.php?id=9006
Q35, spinlock.h error with RHEL 4
please refer this link:
http://forums.digium.com/viewtopic.php?p=17034&sid=c914a0a979f8437576c7aa92518fe48b
Q36, Compile error on CentOS-4.6 with Kernel-2.6.9-67.0.15.ELsmp and CONFIG_DAHDI_NET
please refer this link:
https://issues.asterisk.org/view.php?id=13427
Q37, dahdi_compat.h:31:27: error: zaptel/zaptel.h: No such file
please read this:
https://issues.asterisk.org/view.php?id=14121
Q38, when compiling zaptel/bristuff, error: You do not appear to have the sources for...
please refer these links:
http://forums.digium.com/viewtopic.php?t=7061
http://lists.digium.com/pipermail/asterisk-users/2007-June/189259.html
Q39, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol
please refer this:
http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2007-August/009225.html
Q40, How to install Octasic SoftEcho
please refer these links:
http://www.openvox.cn/download/user_manuals_english_version/Octvqeug_5000.pdf
http://www.octasic.com/en/products/softecho/softecho_asterisk.php
http://www.octasic.com/en/products/softecho/support.php
Q41, Bug in Zaptel 1.2.20.1 and 1.4.5.1 - Only MG2
please refer this:
http://trixbox.org/node/21080 http://www.rowetel.com/ucasterisk/oslec.html
Q42, Howto: OSLEC echo canceling + DAHDI 2.1.0.4 + Asterisk 1.4
please refer this:
http://www.asterisk.org/forum/viewtopic.php?p=125314&sid=9515c7b03cb14dc698e89467c3d49a86
Q43, Difference between zaptel and dahdi
please refer these links:
http://www.voip-info.org/wiki/view/DAHDI
http://docs.tzafrir.org.il/dahdi-linux/
http://docs.tzafrir.org.il/dahdi-tools/
Q44, Tonezones for zaptel.conf
The file zonedata.c contains the information about the tone zones used in libtonezone (and hence also in ztcfg). Here is a list of those zones:
us United States / North America
au Australia
fr France
nl Netherlands
uk United Kingdom
fi Finland
es Spain
jp Japan
no Norway
at Austria
nz New Zealand
it Italy
us-old United States Circa 1950 / North America
gr Greece
tw Taiwan
cl Chile
se Sweden
be Belgium
sg Singapore
il Israel
br Brazil
hu Hungary
lt Lithuania
pl Poland
za South Africa
pt Portugal
ee Estonia
mx Mexico
in India
de Germany
ch Switzerland
dk Denmark
cz Czech Republic
cn China
ar Argentina
my Malaysia
th Thailand
bg Bulgaria
ve Venezuela
ph Philippines
ru Russian Federation
tr Turkey
Q45, Tools from zaptel to dahdi
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool
zapconf -> dahdi_genconf (deprecates genzaptelconf)
Q46, Why are you unable to call out with Asterisk 1.4.22?
If you are using wctdm or opvxa1200 with Zaptel and Asterisk 1.4.22 then there is a known issue with outbound calls. The reason you are not able to call out is because Asterisk 1.4.22 has a new feature which detects if a analog line is plugged in or not, but this feature only works with Dahdi. So to fix the issue you can do one of the following.
edit the file under /asterisk-1.4.22、channels/chan_dahdi.c" find this line
1. ifdef DAHDI_CHECK_HOOKSTATE return 0;
2. else return 1;
Change the "0" to a "1"
1. ifdef DAHDI_CHECK_HOOKSTATE return 1;
2. else return 1;
Q47, install Asterisk zaphfc
please refer these links:
http://www.voip-info.org/wiki/view/Asterisk+zaphfc+install
http://www.openvox.cn/download/user_manuals_english_version/B100P_User_Manual_bristuff.pdf
Q48, Install bri card and analog cards in one system
if you want to install analog cards and bri cards in one system, it can be done in two ways:
1) install bristuff for both of analog cards and bri cards.
2) install mISDN with bri cards and zaptel with analog cards
please refer this link:
http://www.howtoforge.com/asterisk-zaptel-libpri-misdn-asterisk-addons-asterisk-gui-on-debian-etch
Q49, mISDN v2(LCR) with debian/asterisk
please refer the link:
http://blog.runtux.com/2009/03/09/61/
Q50, patch---mISDN rejects incoming calls
misdn rejects incoming calls, please refer this patch:
https://issues.asterisk.org/view.php?id=13488
Q51, B200P/B400P/B100P with trixbox 2.8 can't load chan_misdn.so
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=890&extra=page%3D1
Q52, How to install B200P/B400P/B100P/B800P with trixbox 2.6
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=1016&page=1&extra=page%3D1
Q53, How to install Ubuntu Server 9.04 with Asterisk and OpenVox B800P
Please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=1025&page=1&extra=page%3D1#pid4369
Q54, what kinds of protocols does mISDN support
mISDN so far, supports DSS1(EURO ISDN), it does not support NI1(National ISDN 1).
Q55, To get the quick support, what information about the system I should send?
* Versions of kernel and Linux distribution
* Versions of asterisk and zaptel/dahdi
* Model of OpenVox products
* Debug and error information from your system and asterisk
* Zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf) configuration files and extension.conf if your have trouble with dialplan
* After loading the driver, run the command: demsg and send the information to us
* Output of the command: cat /proc/interrupts
* Message in asterisk console when you making a call, if you have trouble to drop/receive calls.
* Protocols you want to use(FXS,FXO,PRI..)
* Ssh account with root permission if you need us to check the system.
* Make a backup for your important files
* Describe the problem in details
* Serial numbers of OpenVox's products
Troubleshooting of Analog cards
Q1, You can not compile zaptel and asterisk
please make sure that:
1) You have installed all necessary packages and kernel source.
2) Make sure the version of kernel source is exactly same with the version of the kernel.
please check the few links:
http://wiki.openvox.cn/index.php/A1200P
http://wiki.openvox.cn/index.php/A400P
http://www.asteriskguru.com/tutorials/
3) make sure that you do not miss any packages or files in asterisk or zaptel.
4) make sure your system can access www.asterisk.org.
Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)
please check:
1) run lspci -vvvvv, make sure the system can detect the card. Tiger jet chip will be found. If there is no such Tiger jet chip, please clean the PCI slot and try again.
2) if lspc can find the card, make sure the pci id is included in the PCI table in our driver. how to patch the picid, please refer this link:
http://www.openvox.cn/kb/entry/2/
3) if step 1 and step 2 are ok, please check the zaptel.conf or system.conf to make sure that the setting is correct.
4) if step 3 is correct, please make sure that there is no mISDN tiger jet module in the system, if it is there, please remove that or add to blacklist.
5) if you still can not boot it up, you have to recompile zaptel or dahdi again.
Q3, You can not make calls from asterisk
there are few reasons why you can not make calls:
1) check your extensions from your asterisk side, make sure your sip is ready to make calls, and SIP is with a right context what you put in extensions.conf
2) your wctdm or opvxa1200 does not boot up(leds are off).
3) leds are up and card driver has boot up properly, but the zapata.conf is
, so asterisk does not boot up properly,
please check by run: zap show channels
if is empty or no such command, you should check your zapata.conf
4) You maybe recompile your zaptel and asterisk again.
Q4, How do you adjust the volume of voice for analog cards?
You can edit the zapata.conf and change rxgain=5 and txgain=6 or other values. you can use ztmonitor to test that.check from here:
http://linux.die.net/man/8/ztmonitor
Q5, You can not hangup calls
To resolve the problem, please check:
1) set timezone and defaulzone to your country, set country=your country in indication.conf and run: modprobe wctdm/opvxa1200 opermode=YOUR country
2) open busydetect=yes and busycount=4
3) ask your provider to open the "disconnect supervision" service check for more details,
please go here:
http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html
Q6, You can not get the callerid
If you have a problem with callerid, please check with this link:
http://bbs.openvox.cn/viewthread.php?tid=831&extra=page%3D1
or edit the line:
in asterisk/main/dsp.c,
========================
1. define DTMF_THRESHOLD 8.0e7
2. define FAX_THRESHOLD 8.0e7
3. define FAX_2ND_HARMONIC 2.0 /* 4dB */
4. define RADIO_RELAX // add this line
5. ifdef RADIO_RELAX
6. define DTMF_NORMAL_TWIST
=========================
after editing, recompiling asterisk.
Q7, Call conversation suddenly dropped
please refer this reference from digium:
Dropped Calls on TDM
If you are having dropped calls on a TDM400P card or an X100P card there are several things that might cause this.
1)BusyDetect
2)CallProgress
BusyDetect and CallProgress may cause Asterisk to detect false hangups. Setting BusyCount to a higher value or turning off CallProgress may fix the problem. An excessive number of IRQMisses may also cause these problems.
link:http://kb.digium.com/entry/71/
3) try to set
/etc/modprobe.conf，install wctdm to：
install wctdm /sbin/modprobe --ignore-install wctdm battdebounce=128 && /sbin/ztcfg
Q8, How can you set the analog card for your country?
To set asterisk pbx with your country support, you must:
1) set timezone and defaultzone to your country in zaptel.conf or system.conf of dahdi
2) set the country=your country in indication.conf
3) modprobe wctdm or opvxa1200 opermode=YOUR country with capital letter.
4) cat the opermode to confirm the parameter has been loaded
[root@bogon misc]# cat /sys/module/wctdm/parameters/opermode
CHINA
[root@bogon misc]#
if using opvxa1200, please run in this way:
cat /sys/module/opvxa1200/parameters/opermode
5) after load the drivers, run dmesg command to check the mode.
Q9, How can you open the debug for asterisk?
1) You can edit the file logger.conf under /etc/asterisk,
enable the debug or error, those message will be stored under
/var/log/asterisk
2) you also can start your asterisk in this way:
asterisk -vvvvvvvvgc -d
Q10, How can i check the IRQ of analog cards?
please run the command:
cat /proc/interrupts
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.
more details, please check from here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Q11, Where are the opvxa1200 drivers/user manuals for dahdi and zaptel?
Under the download, you can see that there are three subdirectories:
First one is driver, you can get the individual opvxa1200 driver.
Second is a zaptel with opvxa1200, you can choose a proper version for you.
Third one is for dahdi, if you want to try dahdi, you can download whole packages.
link: http://www.openvox.cn/download/
Q12, Sound Quality Problems with Analog cards
please refer this link:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Q13, How can you compile asterisk with dahdi for wctdm and opvxa1200
please refer these links:
http://bbs.openvox.cn/viewthread.php?tid=574&extra=page%3D3
http://bbs.openvox.cn/viewthread.php?tid=587&extra=page%3D1
http://www.openvox.cn/download/
http://www.voip-info.org/wiki/view/DAHDI
http://www.russellbryant.net/blog/category/dahdi/
http://blog.paulsnet.org/?p=44
http://docs.tzafrir.org.il/dahdi-tools/?C=S%3BO=A
Q14, I am hearing an echo. What can I do to fix this?
please refer these links:
http://kb.digium.com/entry/1/
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
Q15, Asterisk does not properly detect when a caller hangs up the phone. How do I fix this?
please refer this link:
http://kb.digium.com/entry/6/
Q16, When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP?
For the TDM400P and TE110P cards, the LED's will not be lit up until the kernel module is loaded. The TDM400P LED's will light up when the ports are configured and the kernel module is loaded. They do not change if a phone or trunk is plugged in or not. The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is connected the LED should turn GREEN.
For the TE2XXP/TE4XXP the LED's should scroll(knightrider) RED even without the kernel module being loaded or anything plugged in. When you have the spans properly configured and kernel module loaded without a circuit or channel bank the LED's should pulse RED. With the module loaded and a circuit/channel bank connected they should be solid GREEN. link from here:
http://kb.digium.com/entry/13/
Q17, What are the differences between FXS and FXO interfaces?
FXS (Foreign eXchange Station) is an interface which drives a telephone. FXS interfaces get phones plugged into them, delivery battery, and provide ringing. FXS interfaces are signalled with FXO signalling.
FXO (Foreign eXchange Office) is an interface that connect to a phone line. They supply your PBX with access to the public telephone network. FXO interfaces use FXS signalling. FXS interfaces are what allow you to hook telephones to your PBX, and FXO interfaces allow you to connect your PBX to real analog phone lines.
Q18, What is the difference between loopstart, groundstart, and kewlstart signalling?
Loopstart signalling is used by virtually all analog phone lines. It allows a phone to indicate on hook/offhook, and the switch to indicate ring/no ring.
Kewlstart is based on loopstart, but extends the protocol by allowing the switch to drop battery on the phone line to indicate to the phone that the other end of the party has disconnected the call. Most real phone switches, and almost no PBX's (except Asterisk, of course) support this feature. It is generally required for getting hangup notification.
Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it.
Q19, Why is my card getting an IRQ miss?
Each peice of hardware takes 1,000 interrupts per second. When, for some reason the cards get less than this, an IRQ miss occurs. You can see if the card is missing interrupts using 'zttool.'
IRQ misses can cause different problems with Asterisk. Symptoms of IRQ misses are bad audio quality or perhaps PRI errors, although IRQ misses will not cause alarms. Also DTMF detection not working is something that can be caused by IRQ misses as well.
Several common things that contribute to IRQ misses are: -Running the X window system -Shared IRQs -No hard drive DMA -Hard drive DMA too high (shoot for udma3) -Running serial terminals or frame buffers
To check for shared IRQs you can run:
1. cat /proc/interrupts
CPU0
0 10756672 XT-PIC timer 2 0 XT-PIC cascade 5 10812879 XT-PIC uhci_hcd, uhci_hcd, wctdm 10 226219 XT-PIC t1xxp, CS46XX 11 1550046 XT-PIC eth0, nvidia 12 387234 XT-PIC i8042 14 32641 XT-PIC ide0 15 18 XT-PIC ide1 NMI 0 LOC 10757616 ERR 40481 MIS 0
Notice the T100P card sharing with the sound card, and the TDM400P card is sharing with the USB controller. This will most likely cause problems. If you are not using any USB devices that would probably be ok, but it would be best to disable USB or get the card on it's own IRQ.
There are several ways to move cards to their own IRQ.
-Turn on APIC
-Tweak BIOS settings
-Try a different PCI slot
-Use setpci
refer this link from digium: http://kb.digium.com/entry/63/
Q20, What should I do if my FXS fails calibration?
Try compiling the kernel without frame buffer support.
link:http://kb.digium.com/entry/61/
Q21, Why am I having DTMF detection problems?
Zaptel DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf. It may need to be turned on or off. If you need to force all DTMF detection to be done in software, you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime.
SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.
Q22, I am getting error messages about PCI Master Aborts. What is wrong?
This is a very rare case. When your computer's PCI subsystem experiences serious problems with OpenVox's cards upon initialization of the card, Linux will print out scrolling "PCI Master Abort" messages. What you should do is go into your system's BIOS, and turn off your motherboard's PNP (plug and play) feature. If this does not resolve your issue, You should contact OpenVox support.
Q23, Why is there a pause after the last DTMF digit?
If you are experiencing a delay or pause before the last DTMF digit is dialed on a Zaptel line, this is because you have echotraining enabled in your zapata.conf. The echotraining is done just before the last digit is dialed, thus the reason for the pause. To fix this you can either set a lower value for echotraining or turn it off completely.
Q24, Why am I getting a clicking noise?
If a clicking noise is present when dialing through an FXO or when getting dialtone from an FXS, this is cause by echotraining. Turn it off to get rid of the clicking. The click is necessary for the echotraining.
Q25, list of asterisk pbx distributions:
www.elastix.org
www.trixobx.org
Q26, How can you install asterisk with Debian Ubutun
http://www.debianhelp.co.uk/asterisk.htm
http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu
http://ubuntuforums.org/showthread.php?t=136785
Q27, How can you install asterisk with Fedora?
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
http://www.asteriskguru.com/
Q28, How can you install asterisk with SuSe?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_suse.html
http://voip-manager.net/installation-linux-asterisk.php
Q29, install asterisk with Free BSD
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
http://www.voip-info.org/wiki/view/FreeBSD+zaptel
Q30, List of Asterisk OS Platforms
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms
Q31, Centos with asterisk
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos
Q32, A1200P "TRUNK Dial failed due to CONGESTION" Problem
please refer this link:
http://www.openvox.cn/bbs/viewthread.php?tid=739&extra=page%3D1
Q33, A1200P Installation on Ubuntu 8.10 Server
if you have a installation problem with A1200P/A800P,
please refer this:http://bbs.openvox.cn/viewthread.php?tid=819&extra=page%3D1
Q34, Can't retrieve Taiwan's CID
please refer this for your problem:
link:http://bbs.openvox.cn/viewthread.php?tid=726&extra=page%3D3
Q35, FXO sends the digits of speed
if you want to add speed dialup for FXO, please refer this:
http://bbs.openvox.cn/viewthread.php?tid=666&extra=page%3D3
Q36, Trick to solve buffer re-sync issue of A1200P/A800P
if your system keeps flushing the "buffer re-sync problems"
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=405&extra=page%3D6
Q37, How do you report a problem
In order to solve customer's problems very effectively and efficiently,
when seeking a help from us, please give us:
1) versions of kernel and Linux distribution
2) versions of asterisk and zaptel/dahdi
3) the name of cards used in your system
4) debug and error information from your system and asterisk
5) sending us zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf)
configuration files and extension.conf
6) after loading the driver, run the command: demsg and send the information to us
7) sending us the result of the command: cat /proc/interrupts
8) sending us the message of asterisk console when you making a call
9) inform the protocols you are using in your system
10) send us a working ssh account with root permission if you need us to check the system.
11) make a backup for your important files
12) describe the problem in details
13) sending the serial numbers of OpenVox's cards
Q38,bug: FXO can not call out!
The problem is this:
1) I cannot make outbound calls on an FXO card (Using FXSKS signalling)until I've received an incoming call
2) resetting the hookstate to offhook(A single ring is good enough),
or alternatively, disconnecting and reconnecting the telephone line.
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=740&extra=page%3D2
Q39,FATAL: Module wctdm not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the wctdm.ko is under /lib/modules/2.6.XX/extra
Q40,FATAL: Module opvxa1200 not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the opvxa1200.ko is under /lib/modules/2.6.XX/extra
Q41,Tools for wctdm and opvxa1200
you can use these tools to test the wctdm and opvxa1200
1) zttest
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
2) zttool
http://www.voip-info.org/wiki/view/Asterisk+zttool
3) ztmonitor
http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment
4) fxotune
http://www.voip-info.org/wiki/view/Asterisk+fxotune
http://kb.digium.com/entry/134/
Q42,check information of wctdm.ko/opvxa1200.ko
Under /lib/modules/2.6.18-128.el5/misc
run command: modinfo wctdm.ko, run modinfo opvxa1200 if checking opvxa1200
[root@bogon misc]# modinfo wctdm.ko
filename: wctdm.ko
license: GPL
alias: wcfxs
author: Mark Spencer
description: Wildcard TDM400P Zaptel Driver
srcversion: 5E22C66ED4D5B1ADE573C83
alias: pci:v0000E159d00000001sv0000A901sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A908sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A801sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A800sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A8FDsd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A9FDsd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B119sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B118sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B1D9sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B100sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000E159sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A159sd*bc*sc*i* depends: zaptel
vermagic: 2.6.18-128.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1
parm: debug:int
parm: loopcurrent:int
parm: reversepolarity:int
parm: robust:int
parm: opermode:charp
parm: timingonly:int
parm: lowpower:int
parm: boostringer:int
parm: fastringer:int
parm: fxshonormode:int
parm: battdebounce:uint
parm: battalarm:uint
parm: battthresh:uint
parm: ringdebounce:int
parm: fwringdetect:int
parm: alawoverride:int
parm: fastpickup:int
parm: fxotxgain:int
parm: fxorxgain:int
parm: fxstxgain:int
parm: fxsrxgain:int
[root@bogon misc]# pwd
/lib/modules/2.6.18-128.el5/misc
Q43, How to debug wctdm or opvxa1200
When loading the wctdm or opvxa1200 with a debug mode,
please loadding the driver in this way:
modprobe wctdm debug=1 // open the debug and check the /var/log/message
Q44, RHEL/Centos 5.2: xpp/xdefs.h:117: error: conflicting types for ‘bool’
please refer this:
https://issues.asterisk.org/view.php?id=12889
Q45, xpp modules do not compile with kernel 2.6.19-1.2919.fc7
please refer this:
https://issues.asterisk.org/view.php?id=9006
Q46, spinlock.h error with RHEL 4
please refer this link:
http://forums.digium.com/viewtopic.php?p=17034&sid=c914a0a979f8437576c7aa92518fe48b
Q47, Compile error on CentOS-4.6 with Kernel-2.6.9-67.0.15.ELsmp and CONFIG_DAHDI_NET
please refer this link:
https://issues.asterisk.org/view.php?id=13427
Q48, dahdi_compat.h:31:27: error: zaptel/zaptel.h: No such file
please read this:
https://issues.asterisk.org/view.php?id=14121
Q49, when compiling zaptel, error: You do not appear to have the sources for...
please refer this:
http://forums.digium.com/viewtopic.php?t=7061
http://lists.digium.com/pipermail/asterisk-users/2007-June/189259.html
Q50, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol
please refer this:
http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2007-August/009225.html
Q51, DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
please check from here:
http://www.openvox.com.cn/bbs/viewtopic.php?t=538&sid=cae3adbe99e80f500d9c9ea7edb52bfb
http://www.openvox.cn/download/other_docs/A800P_A1200P_FAQ(Chinese).pdf
Q52, How to install Octasic SoftEcho
please refer these links:
http://www.openvox.cn/download/user_manuals_english_version/Octvqeug_5000.pdf
http://www.octasic.com/en/products/softecho/softecho_asterisk.php
http://www.octasic.com/en/products/softecho/support.php
Q53, Bug in Zaptel 1.2.20.1 and 1.4.5.1 - Only MG2
please refer this:
http://trixbox.org/node/21080 http://www.rowetel.com/ucasterisk/oslec.html
Q54, Howto: OSLEC echo canceling + DAHDI 2.1.0.4 + Asterisk 1.4
please refer this:
http://www.asterisk.org/forum/viewtopic.php?p=125314&sid=9515c7b03cb14dc698e89467c3d49a86
Q55, Difference between zaptel and dahdi
please refer these links:
http://www.voip-info.org/wiki/view/DAHDI
http://docs.tzafrir.org.il/dahdi-linux/
http://docs.tzafrir.org.il/dahdi-tools/
Q56, Tonezones for wctdm and opvxa1200
The file zonedata.c contains the information about the tone zones used in libtonezone (and hence also in ztcfg). Here is a list of those zones:
us United States / North America
au Australia
fr France
nl Netherlands
uk United Kingdom
fi Finland
es Spain
jp Japan
no Norway
at Austria
nz New Zealand
it Italy
us-old United States Circa 1950 / North America
gr Greece
tw Taiwan
cl Chile
se Sweden
be Belgium
sg Singapore
il Israel
br Brazil
hu Hungary
lt Lithuania
pl Poland
za South Africa
pt Portugal
ee Estonia
mx Mexico
in India
de Germany
ch Switzerland
dk Denmark
cz Czech Republic
cn China
ar Argentina
my Malaysia
th Thailand
bg Bulgaria
ve Venezuela
ph Philippines
ru Russian Federation
tr Turkey
Q57, Tools from zaptel to dahdi
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool
zapconf -> dahdi_genconf (deprecates genzaptelconf)
Q58, the list of opermode
when loading the driver wctdm/opvxa1200,
modprobe wctdm opermode=YOUR COUNTRY
please check from the list: fxo_mudules.h
{
US, Canada
{ .name = "FCC",
.rt = 1,
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
/* Austria, Belgium, Denmark, Finland, France, Germany,
Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
Norway, Portugal, Spain, Sweden, Switzerland, and UK */
{ .name = "TBR21",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.ring_osc = 0x7e6c,
.ring_x = 0x023a,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "ARGENTINA",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "AUSTRALIA",
.ohs = 1,
.mini = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "AUSTRIA",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "BAHRAIN",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "BELGIUM",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "BRAZIL",
.mini = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "BULGARIA",
.ilim = 1,
.dcv = 0x3,
.mini = 0x0,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CANADA",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CHILE",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CHINA",
.mini = 0x3,
.acim = 0xf,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "COLOMBIA",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "CROATIA",
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "CYPRUS",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CZECH",
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "DENMARK",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ECUADOR",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "EGYPT",
.mini = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ELSALVADOR",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "FINLAND",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "FRANCE",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GERMANY",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GREECE",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GUAM",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000, .battthresh = 3,
}, { .name = "HONGKONG",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "HUNGARY",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ICELAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "INDIA",
.dcv = 0x3, .acim = 0x4, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "INDONESIA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000,
.battthresh = 3,
}, { .name = "IRELAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ISRAEL",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ITALY",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "JAPAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "JORDAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "KAZAKHSTAN",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "KUWAIT",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LATVIA",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LEBANON",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LUXEMBOURG",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MACAO",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, /* Current loop >= 20ma */
{ .name = "MALAYSIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MALTA",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MEXICO",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MOROCCO",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NETHERLANDS",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
},
{ .name = "NEWZEALAND",
.dcv = 0x3, .acim = 0x4, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NIGERIA",
.ilim = 0x1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NORWAY",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "OMAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PAKISTAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PERU",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PHILIPPINES",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "POLAND",
.rz = 1, .rt = 1, .dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PORTUGAL",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ROMANIA",
.dcv = 3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "RUSSIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SAUDIARABIA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SINGAPORE",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SLOVAKIA",
.dcv = 0x3, .acim = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SLOVENIA",
.dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SOUTHAFRICA",
.ohs = 1, .rz = 1, .dcv = 0x3, .acim = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SOUTHKOREA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SPAIN",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SWEDEN",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SWITZERLAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SYRIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "TAIWAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "THAILAND",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "UAE",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "UK",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x5, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "USA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "YEMEN",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, };
Q59, Callerid with DTMF and FSK
please refer these links:
http://www.ainslie.org.uk/callerid/cli_faq.htm
http://www.asteriskguru.com/tutorials/calleridall_function.html
https://issues.asterisk.org/view.php?id=7787
http://asterisk.pbx.in/callerid-standards
http://kb.digium.com/entry/3/
Q60, Why are you unable to call out with Asterisk 1.4.22?
If you are using wctdm or opvxa1200 with Zaptel and Asterisk 1.4.22 then there is a known issue with outbound calls. The reason you are not able to call out is because Asterisk 1.4.22 has a new feature which detects if a analog line is plugged in or not, but this feature only works with Dahdi. So to fix the issue you can do one of the following.
edit the file under /asterisk-1.4.22、channels/chan_dahdi.c" find this line
1. ifdef DAHDI_CHECK_HOOKSTATE return 0;
2. else return 1;
Change the "0" to a "1"
1. ifdef DAHDI_CHECK_HOOKSTATE return 1;
2. else return 1;
Q61, Sound quality issue with wctdm and opvxa1200
you can try these possible solutions:
1) Check the IRQ, make sure system handle IRQ properly
2) Use zttest to check the result
3) Disable the X window and framebuffer(set vga=normal from grub)
4) When using IDE driver, please open the DMA control
5) Bind the card IRQ to particular CPU, for example:
echo 1 > /proc/irq/217/smp_affinity #wcfxo
6) Set the latency timer: setpci -v -s 04:06.0 LATENCY_TIMER=f8 #wcfxo
Q62, Echotraining and OSLEC
if you use the OSLEC, you must disable the echotraining.
otherwise, the channel will be silent.
Q63, Far side disconnected with disconnect supervisions
possible solutions:
1) Your PSTN provider uses voltage/battery drop for disconnect supervision,
pleas try to edit DCT in zaptel.h to 100 ms or higher.
2) If you PSTN provider provides reverse polarity/battery
disconnect supervision:
hanguponpolarityswitch=yes
answeruponpolarityswitch=yes
3) If your PSTN provider has disconnect/busy tone disconnect supervision
busydetect=yes busycount=6
or try to callprogress, and set your progzone to your country code in zapata.conf
Q64, Cabling for wctdm and opvxa1200
wctdm/opvxa1200 uses RJ11 cable. please refer this:
http://www.tech-faq.com/rj-11.shtml
http://en.wikipedia.org/wiki/RJ11,_RJ14,_RJ25
Q65, Splitter for A1200P/A800P
http://openvox.cn/products/show.php?itemid=120&lang=2
Q66, Power supply for for wctdm and opvxa1200
If you use FXS with wctdm/opvxa1200, you MUST use
12v power supply(4 pin Molex power cable) for FXS, please refer this:
http://www.playtool.com/pages/psuconnectors/connectors.html#peripheral
http://en.wikipedia.org/wiki/Computer_power_supply
Q67, Configuring systems for the UK (various country specific quirks)
if users are in UK, please check the cable and callerid set from here: Interfacing with the UK Analogue telephone network This is a how to for connecting your fonicavoip PABX to analogue phone lines in Britain. Our signalling protocols on analogue phone lines date back to the days of the Government owned General Post Office, and are different from those used in America or mainland Europe. Wiring
The UK uses a unique telephone socket called a BS6312 socket or "BT socket", which is different from the RJ11 plug commonly used elsewhere. To connect most line cards to the telephone line you need a BT plug to RJ11 plug connector, where pins 2 and 5 of the BT plug are connected to the inner two pins (2 and 3) of the RJ11 plug. Most phone leads supplied for recent analogue modems are wired in this way but beware - there are two kinds of these leads on the market - one commonly used with older (1990s) BT phones and modems has the phone line on the outer two pins of the RJ11. This will not at all with single line RJ11 sockets found on line cards! Caller ID
On BT lines, Caller ID in the UK is sent before the first ring and after a polarity reversal. These lines must be inserted into your Zaptel or DAHDI conf file so UK caller ID works properly.
usecallerid=yes
cidsignalling=v23
cidstart=polarity Featurelines
These are the UK version of Centrex, and BT often tend to encourage small businesses to take these on on a long term contract with fairly competitive rates. They have the issue that even with a standard phone connected, the trunk requires 9 to be dialled to get an "outside line" as it you were already connected to a PABX - also incoming caller ID is prefixed with this 9.
For an extra charge BT offer "internal extensions" which can be in different sites provided they are connected to the same Telephone Exchange. These usually are two digits long and start with "Extension 20"
With a normal PABX connected to these lines you would need to dial 9 'twice' to get an outside line (or whatever other access code is in use). This is a hassle, and could potentially even lead to false 999 (emergency) calls!
However FreePBX's advanced routing can be used to ensure users only have to dial 9 once for the outside line - and can still make use of the internal extensions - thus getting the best of both worlds.
1. In Trunks
set up your ZAP trunks connected to the Centrex/PABX line for your normal PSTN dial plan but coded to add "9" as if you were dialling from an extension.
For instance
9+999
9+112
9+NXXXXX
9+0.
9+1.
2. In Outbound Routes
Set up 9_outside to point to these trunks
9|.
Now your end users only have to dial "9" once and will get routed to outside line provided it matches your dialplan in trunks.
Now for the Centrex/PABX "extensions":
Choose another number as your "inter-PABX" code. I have picked "8" (as traditionally thats how many British phone systems have been set up).
Set up an outbound route to look for your Centrex/PABX extension dialling pattern after the 8 (remember that my extensions were 20-24)
8|2[01234]
Point this route to the Centrex/PABX trunks on the ZAP channel - and anything prefixed with 8 will be dialled as an "internal" extension.
Incidentally If you don't want to move your users to a longer extension digit length and the Centrex/PABX extensions are all in the same number range you can of course tweak the route to just route all numbers beginning with 2 for instance via this method, provided you don't have any FreePBX extensions in the same number range... BT Line testing and a caveat with current versions of DAHDI (as at 2009-03-15)
Many BT lines are automatically tested every night. This is to test if the cable between the Telephone Exchange and your NTE5 (the line socket) has been damaged. To do this, equipment within the exchange removes the normal dial tone and -50 volts DC (battery) from your line and substitutes it with a variety of DC and AC signals.
This appears to Zap/DAHDI hardware as a series of battery drops and line reversals
Previous versions of zaptel/DAHDI did not check the telephone line for battery voltage. This meant that the reversals picked up from the testing sequence are treated as a a "pre-ring" state (before caller ID) but time out harmlessly, so the channel was only shown unavailable for the duration of the line test (which is a sensible thing to happen).
Newer versions of DAHDI do check the line for battery. This is also sensible, as it stops Asterisk dialling into an unconnected telephone circuit.
Unfortunately, the way Asterisk is currently coded means that if the system is configured for the UK caller ID protocol, any on hook polarity reversal is treated as a precursor to the telephone ringing, when in fact (according to BT's own specifications) line reversals can randomly happen at any time.
Another piece of code inserted to make Asterisk ignore alarms at this time has inadvertently caused the combination of battery drops and reversals to be picked up as a permanent line fault.
This is under investigation at the link below, with a patch available for chan_dahdi.c
http://bugs.digium.com/view.php?id=14163
the original link from: http://www.fonicaprojects.com/wiki/index.php/Configuring_systems_for_the_UK_%28various_country_specific_quirks%29
Q68, User has to make incoming call first, then you can make outgoing calls.
certain versions of asterisk does not detect offhook correctly, there are two ways to use to make your system working as normal status. 1) plug out the fxo cable, then plug back that 2) you have to call in, then you can make outgoing calls. to solve this problem, you have to patch asterisk. this is a patch for asterisk1-6.2.2: --- org_chan_dahdi.c 2010-02-04 21:33:34.000000000 -0500 +++ new_chan_dahdi.c 2010-02-04 21:35:48.000000000 -0500 @@ -198,7 +198,7 @@
* before dialing on it. Certain FXO interfaces always think they're out of
* service with this method however.
*/
-/* #define DAHDI_CHECK_HOOKSTATE */ +#define DAHDI_CHECK_HOOKSTATE
/*! \brief Typically, how many rings before we should send Caller*ID */
1. define DEFAULT_CIDRINGS 1
@@ -10685,12 +10685,16 @@
/* When "onhook" that means no battery on the line, and thus
it is out of service..., if it's on a TDM card... If it's a channel
bank, there is no telling... */
+#ifdef DAHDI_CHECK_HOOKSTATE
if (par.rxbits > -1)
return 1;
if (par.rxisoffhook)
return 1;
else
return 0;
+#else + return 1; +#endif
} else if (par.rxisoffhook) {
ast_debug(1, "Channel %d off hook, can't use\n", p->channel);
/* Not available when the other end is off hook */
more details: https://issues.asterisk.org/file_download.php?file_id=23171&type=bug
Q69, To get the quick support, what information about the system I should send?
* Versions of kernel and Linux distribution
* Versions of asterisk and zaptel/dahdi
* Model of OpenVox products
* Debug and error information from your system and asterisk
* Zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf) configuration files and extension.conf if your have trouble with dialplan
* After loading the driver, run the command: demsg and send the information to us
* Output of the command: cat /proc/interrupts
* Message in asterisk console when you making a call, if you have trouble to drop/receive calls.
* Protocols you want to use(FXS,FXO,PRI..)
* Ssh account with root permission if you need us to check the system.
* Make a backup for your important files
* Describe the problem in details
* Serial numbers of OpenVox's products
please make sure that:
1) You have installed all necessary packages and kernel source.
2) Make sure the version of kernel source is exactly same with the version of the kernel.
please check the few links:
http://wiki.openvox.cn/index.php/A1200P
http://wiki.openvox.cn/index.php/A400P
http://www.asteriskguru.com/tutorials/
3) make sure that you do not miss any packages or files in asterisk or zaptel.
4) make sure your system can access www.asterisk.org.
Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)
please check:
1) run lspci -vvvvv, make sure the system can detect the card. Tiger jet chip will be found. If there is no such Tiger jet chip, please clean the PCI slot and try again.
2) if lspc can find the card, make sure the pci id is included in the PCI table in our driver. how to patch the picid, please refer this link:
http://www.openvox.cn/kb/entry/2/
3) if step 1 and step 2 are ok, please check the zaptel.conf or system.conf to make sure that the setting is correct.
4) if step 3 is correct, please make sure that there is no mISDN tiger jet module in the system, if it is there, please remove that or add to blacklist.
5) if you still can not boot it up, you have to recompile zaptel or dahdi again.
Q3, You can not make calls from asterisk
there are few reasons why you can not make calls:
1) check your extensions from your asterisk side, make sure your sip is ready to make calls, and SIP is with a right context what you put in extensions.conf
2) your wctdm or opvxa1200 does not boot up(leds are off).
3) leds are up and card driver has boot up properly, but the zapata.conf is
, so asterisk does not boot up properly,
please check by run: zap show channels
if is empty or no such command, you should check your zapata.conf
4) You maybe recompile your zaptel and asterisk again.
Q4, How do you adjust the volume of voice for analog cards?
You can edit the zapata.conf and change rxgain=5 and txgain=6 or other values. you can use ztmonitor to test that.check from here:
http://linux.die.net/man/8/ztmonitor
Q5, You can not hangup calls
To resolve the problem, please check:
1) set timezone and defaulzone to your country, set country=your country in indication.conf and run: modprobe wctdm/opvxa1200 opermode=YOUR country
2) open busydetect=yes and busycount=4
3) ask your provider to open the "disconnect supervision" service check for more details,
please go here:
http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html
Q6, You can not get the callerid
If you have a problem with callerid, please check with this link:
http://bbs.openvox.cn/viewthread.php?tid=831&extra=page%3D1
or edit the line:
in asterisk/main/dsp.c,
========================
1. define DTMF_THRESHOLD 8.0e7
2. define FAX_THRESHOLD 8.0e7
3. define FAX_2ND_HARMONIC 2.0 /* 4dB */
4. define RADIO_RELAX // add this line
5. ifdef RADIO_RELAX
6. define DTMF_NORMAL_TWIST
=========================
after editing, recompiling asterisk.
Q7, Call conversation suddenly dropped
please refer this reference from digium:
Dropped Calls on TDM
If you are having dropped calls on a TDM400P card or an X100P card there are several things that might cause this.
1)BusyDetect
2)CallProgress
BusyDetect and CallProgress may cause Asterisk to detect false hangups. Setting BusyCount to a higher value or turning off CallProgress may fix the problem. An excessive number of IRQMisses may also cause these problems.
link:http://kb.digium.com/entry/71/
3) try to set
/etc/modprobe.conf，install wctdm to：
install wctdm /sbin/modprobe --ignore-install wctdm battdebounce=128 && /sbin/ztcfg
Q8, How can you set the analog card for your country?
To set asterisk pbx with your country support, you must:
1) set timezone and defaultzone to your country in zaptel.conf or system.conf of dahdi
2) set the country=your country in indication.conf
3) modprobe wctdm or opvxa1200 opermode=YOUR country with capital letter.
4) cat the opermode to confirm the parameter has been loaded
[root@bogon misc]# cat /sys/module/wctdm/parameters/opermode
CHINA
[root@bogon misc]#
if using opvxa1200, please run in this way:
cat /sys/module/opvxa1200/parameters/opermode
5) after load the drivers, run dmesg command to check the mode.
Q9, How can you open the debug for asterisk?
1) You can edit the file logger.conf under /etc/asterisk,
enable the debug or error, those message will be stored under
/var/log/asterisk
2) you also can start your asterisk in this way:
asterisk -vvvvvvvvgc -d
Q10, How can i check the IRQ of analog cards?
please run the command:
cat /proc/interrupts
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.
more details, please check from here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Q11, Where are the opvxa1200 drivers/user manuals for dahdi and zaptel?
Under the download, you can see that there are three subdirectories:
First one is driver, you can get the individual opvxa1200 driver.
Second is a zaptel with opvxa1200, you can choose a proper version for you.
Third one is for dahdi, if you want to try dahdi, you can download whole packages.
link: http://www.openvox.cn/download/
Q12, Sound Quality Problems with Analog cards
please refer this link:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Q13, How can you compile asterisk with dahdi for wctdm and opvxa1200
please refer these links:
http://bbs.openvox.cn/viewthread.php?tid=574&extra=page%3D3
http://bbs.openvox.cn/viewthread.php?tid=587&extra=page%3D1
http://www.openvox.cn/download/
http://www.voip-info.org/wiki/view/DAHDI
http://www.russellbryant.net/blog/category/dahdi/
http://blog.paulsnet.org/?p=44
http://docs.tzafrir.org.il/dahdi-tools/?C=S%3BO=A
Q14, I am hearing an echo. What can I do to fix this?
please refer these links:
http://kb.digium.com/entry/1/
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
Q15, Asterisk does not properly detect when a caller hangs up the phone. How do I fix this?
please refer this link:
http://kb.digium.com/entry/6/
Q16, When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP?
For the TDM400P and TE110P cards, the LED's will not be lit up until the kernel module is loaded. The TDM400P LED's will light up when the ports are configured and the kernel module is loaded. They do not change if a phone or trunk is plugged in or not. The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is connected the LED should turn GREEN.
For the TE2XXP/TE4XXP the LED's should scroll(knightrider) RED even without the kernel module being loaded or anything plugged in. When you have the spans properly configured and kernel module loaded without a circuit or channel bank the LED's should pulse RED. With the module loaded and a circuit/channel bank connected they should be solid GREEN. link from here:
http://kb.digium.com/entry/13/
Q17, What are the differences between FXS and FXO interfaces?
FXS (Foreign eXchange Station) is an interface which drives a telephone. FXS interfaces get phones plugged into them, delivery battery, and provide ringing. FXS interfaces are signalled with FXO signalling.
FXO (Foreign eXchange Office) is an interface that connect to a phone line. They supply your PBX with access to the public telephone network. FXO interfaces use FXS signalling. FXS interfaces are what allow you to hook telephones to your PBX, and FXO interfaces allow you to connect your PBX to real analog phone lines.
Q18, What is the difference between loopstart, groundstart, and kewlstart signalling?
Loopstart signalling is used by virtually all analog phone lines. It allows a phone to indicate on hook/offhook, and the switch to indicate ring/no ring.
Kewlstart is based on loopstart, but extends the protocol by allowing the switch to drop battery on the phone line to indicate to the phone that the other end of the party has disconnected the call. Most real phone switches, and almost no PBX's (except Asterisk, of course) support this feature. It is generally required for getting hangup notification.
Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it.
Q19, Why is my card getting an IRQ miss?
Each peice of hardware takes 1,000 interrupts per second. When, for some reason the cards get less than this, an IRQ miss occurs. You can see if the card is missing interrupts using 'zttool.'
IRQ misses can cause different problems with Asterisk. Symptoms of IRQ misses are bad audio quality or perhaps PRI errors, although IRQ misses will not cause alarms. Also DTMF detection not working is something that can be caused by IRQ misses as well.
Several common things that contribute to IRQ misses are: -Running the X window system -Shared IRQs -No hard drive DMA -Hard drive DMA too high (shoot for udma3) -Running serial terminals or frame buffers
To check for shared IRQs you can run:
1. cat /proc/interrupts
CPU0
0 10756672 XT-PIC timer 2 0 XT-PIC cascade 5 10812879 XT-PIC uhci_hcd, uhci_hcd, wctdm 10 226219 XT-PIC t1xxp, CS46XX 11 1550046 XT-PIC eth0, nvidia 12 387234 XT-PIC i8042 14 32641 XT-PIC ide0 15 18 XT-PIC ide1 NMI 0 LOC 10757616 ERR 40481 MIS 0
Notice the T100P card sharing with the sound card, and the TDM400P card is sharing with the USB controller. This will most likely cause problems. If you are not using any USB devices that would probably be ok, but it would be best to disable USB or get the card on it's own IRQ.
There are several ways to move cards to their own IRQ.
-Turn on APIC
-Tweak BIOS settings
-Try a different PCI slot
-Use setpci
refer this link from digium: http://kb.digium.com/entry/63/
Q20, What should I do if my FXS fails calibration?
Try compiling the kernel without frame buffer support.
link:http://kb.digium.com/entry/61/
Q21, Why am I having DTMF detection problems?
Zaptel DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf. It may need to be turned on or off. If you need to force all DTMF detection to be done in software, you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime.
SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.
Q22, I am getting error messages about PCI Master Aborts. What is wrong?
This is a very rare case. When your computer's PCI subsystem experiences serious problems with OpenVox's cards upon initialization of the card, Linux will print out scrolling "PCI Master Abort" messages. What you should do is go into your system's BIOS, and turn off your motherboard's PNP (plug and play) feature. If this does not resolve your issue, You should contact OpenVox support.
Q23, Why is there a pause after the last DTMF digit?
If you are experiencing a delay or pause before the last DTMF digit is dialed on a Zaptel line, this is because you have echotraining enabled in your zapata.conf. The echotraining is done just before the last digit is dialed, thus the reason for the pause. To fix this you can either set a lower value for echotraining or turn it off completely.
Q24, Why am I getting a clicking noise?
If a clicking noise is present when dialing through an FXO or when getting dialtone from an FXS, this is cause by echotraining. Turn it off to get rid of the clicking. The click is necessary for the echotraining.
Q25, list of asterisk pbx distributions:
www.elastix.org
www.trixobx.org
Q26, How can you install asterisk with Debian Ubutun
http://www.debianhelp.co.uk/asterisk.htm
http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu
http://ubuntuforums.org/showthread.php?t=136785
Q27, How can you install asterisk with Fedora?
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
http://www.asteriskguru.com/
Q28, How can you install asterisk with SuSe?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_suse.html
http://voip-manager.net/installation-linux-asterisk.php
Q29, install asterisk with Free BSD
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
http://www.voip-info.org/wiki/view/FreeBSD+zaptel
Q30, List of Asterisk OS Platforms
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms
Q31, Centos with asterisk
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos
Q32, A1200P "TRUNK Dial failed due to CONGESTION" Problem
please refer this link:
http://www.openvox.cn/bbs/viewthread.php?tid=739&extra=page%3D1
Q33, A1200P Installation on Ubuntu 8.10 Server
if you have a installation problem with A1200P/A800P,
please refer this:http://bbs.openvox.cn/viewthread.php?tid=819&extra=page%3D1
Q34, Can't retrieve Taiwan's CID
please refer this for your problem:
link:http://bbs.openvox.cn/viewthread.php?tid=726&extra=page%3D3
Q35, FXO sends the digits of speed
if you want to add speed dialup for FXO, please refer this:
http://bbs.openvox.cn/viewthread.php?tid=666&extra=page%3D3
Q36, Trick to solve buffer re-sync issue of A1200P/A800P
if your system keeps flushing the "buffer re-sync problems"
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=405&extra=page%3D6
Q37, How do you report a problem
In order to solve customer's problems very effectively and efficiently,
when seeking a help from us, please give us:
1) versions of kernel and Linux distribution
2) versions of asterisk and zaptel/dahdi
3) the name of cards used in your system
4) debug and error information from your system and asterisk
5) sending us zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf)
configuration files and extension.conf
6) after loading the driver, run the command: demsg and send the information to us
7) sending us the result of the command: cat /proc/interrupts
8) sending us the message of asterisk console when you making a call
9) inform the protocols you are using in your system
10) send us a working ssh account with root permission if you need us to check the system.
11) make a backup for your important files
12) describe the problem in details
13) sending the serial numbers of OpenVox's cards
Q38,bug: FXO can not call out!
The problem is this:
1) I cannot make outbound calls on an FXO card (Using FXSKS signalling)until I've received an incoming call
2) resetting the hookstate to offhook(A single ring is good enough),
or alternatively, disconnecting and reconnecting the telephone line.
please refer this link:
http://bbs.openvox.cn/viewthread.php?tid=740&extra=page%3D2
Q39,FATAL: Module wctdm not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the wctdm.ko is under /lib/modules/2.6.XX/extra
Q40,FATAL: Module opvxa1200 not found
if this problem occurred, please make sure:
1) the module is compiled and installed properly
2) you entered a right kernel, which you used to compile the zaptel
3) make sure you have a access permission to load the module.
4) make sure the opvxa1200.ko is under /lib/modules/2.6.XX/extra
Q41,Tools for wctdm and opvxa1200
you can use these tools to test the wctdm and opvxa1200
1) zttest
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
2) zttool
http://www.voip-info.org/wiki/view/Asterisk+zttool
3) ztmonitor
http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment
4) fxotune
http://www.voip-info.org/wiki/view/Asterisk+fxotune
http://kb.digium.com/entry/134/
Q42,check information of wctdm.ko/opvxa1200.ko
Under /lib/modules/2.6.18-128.el5/misc
run command: modinfo wctdm.ko, run modinfo opvxa1200 if checking opvxa1200
[root@bogon misc]# modinfo wctdm.ko
filename: wctdm.ko
license: GPL
alias: wcfxs
author: Mark Spencer
description: Wildcard TDM400P Zaptel Driver
srcversion: 5E22C66ED4D5B1ADE573C83
alias: pci:v0000E159d00000001sv0000A901sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A908sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A801sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A800sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A8FDsd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A9FDsd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B119sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B118sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B1D9sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000B100sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000E159sd*bc*sc*i* alias: pci:v0000E159d00000001sv0000A159sd*bc*sc*i* depends: zaptel
vermagic: 2.6.18-128.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1
parm: debug:int
parm: loopcurrent:int
parm: reversepolarity:int
parm: robust:int
parm: opermode:charp
parm: timingonly:int
parm: lowpower:int
parm: boostringer:int
parm: fastringer:int
parm: fxshonormode:int
parm: battdebounce:uint
parm: battalarm:uint
parm: battthresh:uint
parm: ringdebounce:int
parm: fwringdetect:int
parm: alawoverride:int
parm: fastpickup:int
parm: fxotxgain:int
parm: fxorxgain:int
parm: fxstxgain:int
parm: fxsrxgain:int
[root@bogon misc]# pwd
/lib/modules/2.6.18-128.el5/misc
Q43, How to debug wctdm or opvxa1200
When loading the wctdm or opvxa1200 with a debug mode,
please loadding the driver in this way:
modprobe wctdm debug=1 // open the debug and check the /var/log/message
Q44, RHEL/Centos 5.2: xpp/xdefs.h:117: error: conflicting types for ‘bool’
please refer this:
https://issues.asterisk.org/view.php?id=12889
Q45, xpp modules do not compile with kernel 2.6.19-1.2919.fc7
please refer this:
https://issues.asterisk.org/view.php?id=9006
Q46, spinlock.h error with RHEL 4
please refer this link:
http://forums.digium.com/viewtopic.php?p=17034&sid=c914a0a979f8437576c7aa92518fe48b
Q47, Compile error on CentOS-4.6 with Kernel-2.6.9-67.0.15.ELsmp and CONFIG_DAHDI_NET
please refer this link:
https://issues.asterisk.org/view.php?id=13427
Q48, dahdi_compat.h:31:27: error: zaptel/zaptel.h: No such file
please read this:
https://issues.asterisk.org/view.php?id=14121
Q49, when compiling zaptel, error: You do not appear to have the sources for...
please refer this:
http://forums.digium.com/viewtopic.php?t=7061
http://lists.digium.com/pipermail/asterisk-users/2007-June/189259.html
Q50, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol
please refer this:
http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2007-August/009225.html
Q51, DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
please check from here:
http://www.openvox.com.cn/bbs/viewtopic.php?t=538&sid=cae3adbe99e80f500d9c9ea7edb52bfb
http://www.openvox.cn/download/other_docs/A800P_A1200P_FAQ(Chinese).pdf
Q52, How to install Octasic SoftEcho
please refer these links:
http://www.openvox.cn/download/user_manuals_english_version/Octvqeug_5000.pdf
http://www.octasic.com/en/products/softecho/softecho_asterisk.php
http://www.octasic.com/en/products/softecho/support.php
Q53, Bug in Zaptel 1.2.20.1 and 1.4.5.1 - Only MG2
please refer this:
http://trixbox.org/node/21080 http://www.rowetel.com/ucasterisk/oslec.html
Q54, Howto: OSLEC echo canceling + DAHDI 2.1.0.4 + Asterisk 1.4
please refer this:
http://www.asterisk.org/forum/viewtopic.php?p=125314&sid=9515c7b03cb14dc698e89467c3d49a86
Q55, Difference between zaptel and dahdi
please refer these links:
http://www.voip-info.org/wiki/view/DAHDI
http://docs.tzafrir.org.il/dahdi-linux/
http://docs.tzafrir.org.il/dahdi-tools/
Q56, Tonezones for wctdm and opvxa1200
The file zonedata.c contains the information about the tone zones used in libtonezone (and hence also in ztcfg). Here is a list of those zones:
us United States / North America
au Australia
fr France
nl Netherlands
uk United Kingdom
fi Finland
es Spain
jp Japan
no Norway
at Austria
nz New Zealand
it Italy
us-old United States Circa 1950 / North America
gr Greece
tw Taiwan
cl Chile
se Sweden
be Belgium
sg Singapore
il Israel
br Brazil
hu Hungary
lt Lithuania
pl Poland
za South Africa
pt Portugal
ee Estonia
mx Mexico
in India
de Germany
ch Switzerland
dk Denmark
cz Czech Republic
cn China
ar Argentina
my Malaysia
th Thailand
bg Bulgaria
ve Venezuela
ph Philippines
ru Russian Federation
tr Turkey
Q57, Tools from zaptel to dahdi
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool
zapconf -> dahdi_genconf (deprecates genzaptelconf)
Q58, the list of opermode
when loading the driver wctdm/opvxa1200,
modprobe wctdm opermode=YOUR COUNTRY
please check from the list: fxo_mudules.h
{
US, Canada
{ .name = "FCC",
.rt = 1,
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
/* Austria, Belgium, Denmark, Finland, France, Germany,
Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
Norway, Portugal, Spain, Sweden, Switzerland, and UK */
{ .name = "TBR21",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.ring_osc = 0x7e6c,
.ring_x = 0x023a,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "ARGENTINA",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "AUSTRALIA",
.ohs = 1,
.mini = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "AUSTRIA",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "BAHRAIN",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "BELGIUM",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "BRAZIL",
.mini = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "BULGARIA",
.ilim = 1,
.dcv = 0x3,
.mini = 0x0,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CANADA",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CHILE",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CHINA",
.mini = 0x3,
.acim = 0xf,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "COLOMBIA",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "CROATIA",
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "CYPRUS",
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "CZECH",
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "DENMARK",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ECUADOR",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "EGYPT",
.mini = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ELSALVADOR",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "FINLAND",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
},
{ .name = "FRANCE",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.mini = 0,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GERMANY",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GREECE",
.ohs2 = 1,
.ilim = 1,
.dcv = 0x3,
.acim = 0x2,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "GUAM",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000, .battthresh = 3,
}, { .name = "HONGKONG",
.dcv = 0x3, .battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "HUNGARY",
.dcv = 0x3,
.battdebounce = 64,
.battalarm = 1000,
.battthresh = 3,
}, { .name = "ICELAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "INDIA",
.dcv = 0x3, .acim = 0x4, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "INDONESIA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000,
.battthresh = 3,
}, { .name = "IRELAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ISRAEL",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ITALY",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "JAPAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "JORDAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "KAZAKHSTAN",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "KUWAIT",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LATVIA",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LEBANON",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "LUXEMBOURG",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MACAO",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, /* Current loop >= 20ma */
{ .name = "MALAYSIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MALTA",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MEXICO",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "MOROCCO",
.ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NETHERLANDS",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
},
{ .name = "NEWZEALAND",
.dcv = 0x3, .acim = 0x4, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NIGERIA",
.ilim = 0x1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "NORWAY",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "OMAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PAKISTAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PERU",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PHILIPPINES",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "POLAND",
.rz = 1, .rt = 1, .dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "PORTUGAL",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "ROMANIA",
.dcv = 3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "RUSSIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SAUDIARABIA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SINGAPORE",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SLOVAKIA",
.dcv = 0x3, .acim = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SLOVENIA",
.dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SOUTHAFRICA",
.ohs = 1, .rz = 1, .dcv = 0x3, .acim = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SOUTHKOREA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SPAIN",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SWEDEN",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SWITZERLAND",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x2, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "SYRIA",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "TAIWAN",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "THAILAND",
.mini = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "UAE",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "UK",
.ohs2 = 1, .ilim = 1, .dcv = 0x3, .acim = 0x5, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "USA",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, { .name = "YEMEN",
.dcv = 0x3, .battdebounce = 64, .battalarm = 1000, .battthresh = 3,
}, };
Q59, Callerid with DTMF and FSK
please refer these links:
http://www.ainslie.org.uk/callerid/cli_faq.htm
http://www.asteriskguru.com/tutorials/calleridall_function.html
https://issues.asterisk.org/view.php?id=7787
http://asterisk.pbx.in/callerid-standards
http://kb.digium.com/entry/3/
Q60, Why are you unable to call out with Asterisk 1.4.22?
If you are using wctdm or opvxa1200 with Zaptel and Asterisk 1.4.22 then there is a known issue with outbound calls. The reason you are not able to call out is because Asterisk 1.4.22 has a new feature which detects if a analog line is plugged in or not, but this feature only works with Dahdi. So to fix the issue you can do one of the following.
edit the file under /asterisk-1.4.22、channels/chan_dahdi.c" find this line
1. ifdef DAHDI_CHECK_HOOKSTATE return 0;
2. else return 1;
Change the "0" to a "1"
1. ifdef DAHDI_CHECK_HOOKSTATE return 1;
2. else return 1;
Q61, Sound quality issue with wctdm and opvxa1200
you can try these possible solutions:
1) Check the IRQ, make sure system handle IRQ properly
2) Use zttest to check the result
3) Disable the X window and framebuffer(set vga=normal from grub)
4) When using IDE driver, please open the DMA control
5) Bind the card IRQ to particular CPU, for example:
echo 1 > /proc/irq/217/smp_affinity #wcfxo
6) Set the latency timer: setpci -v -s 04:06.0 LATENCY_TIMER=f8 #wcfxo
Q62, Echotraining and OSLEC
if you use the OSLEC, you must disable the echotraining.
otherwise, the channel will be silent.
Q63, Far side disconnected with disconnect supervisions
possible solutions:
1) Your PSTN provider uses voltage/battery drop for disconnect supervision,
pleas try to edit DCT in zaptel.h to 100 ms or higher.
2) If you PSTN provider provides reverse polarity/battery
disconnect supervision:
hanguponpolarityswitch=yes
answeruponpolarityswitch=yes
3) If your PSTN provider has disconnect/busy tone disconnect supervision
busydetect=yes busycount=6
or try to callprogress, and set your progzone to your country code in zapata.conf
Q64, Cabling for wctdm and opvxa1200
wctdm/opvxa1200 uses RJ11 cable. please refer this:
http://www.tech-faq.com/rj-11.shtml
http://en.wikipedia.org/wiki/RJ11,_RJ14,_RJ25
Q65, Splitter for A1200P/A800P
http://openvox.cn/products/show.php?itemid=120&lang=2
Q66, Power supply for for wctdm and opvxa1200
If you use FXS with wctdm/opvxa1200, you MUST use
12v power supply(4 pin Molex power cable) for FXS, please refer this:
http://www.playtool.com/pages/psuconnectors/connectors.html#peripheral
http://en.wikipedia.org/wiki/Computer_power_supply
Q67, Configuring systems for the UK (various country specific quirks)
if users are in UK, please check the cable and callerid set from here: Interfacing with the UK Analogue telephone network This is a how to for connecting your fonicavoip PABX to analogue phone lines in Britain. Our signalling protocols on analogue phone lines date back to the days of the Government owned General Post Office, and are different from those used in America or mainland Europe. Wiring
The UK uses a unique telephone socket called a BS6312 socket or "BT socket", which is different from the RJ11 plug commonly used elsewhere. To connect most line cards to the telephone line you need a BT plug to RJ11 plug connector, where pins 2 and 5 of the BT plug are connected to the inner two pins (2 and 3) of the RJ11 plug. Most phone leads supplied for recent analogue modems are wired in this way but beware - there are two kinds of these leads on the market - one commonly used with older (1990s) BT phones and modems has the phone line on the outer two pins of the RJ11. This will not at all with single line RJ11 sockets found on line cards! Caller ID
On BT lines, Caller ID in the UK is sent before the first ring and after a polarity reversal. These lines must be inserted into your Zaptel or DAHDI conf file so UK caller ID works properly.
usecallerid=yes
cidsignalling=v23
cidstart=polarity Featurelines
These are the UK version of Centrex, and BT often tend to encourage small businesses to take these on on a long term contract with fairly competitive rates. They have the issue that even with a standard phone connected, the trunk requires 9 to be dialled to get an "outside line" as it you were already connected to a PABX - also incoming caller ID is prefixed with this 9.
For an extra charge BT offer "internal extensions" which can be in different sites provided they are connected to the same Telephone Exchange. These usually are two digits long and start with "Extension 20"
With a normal PABX connected to these lines you would need to dial 9 'twice' to get an outside line (or whatever other access code is in use). This is a hassle, and could potentially even lead to false 999 (emergency) calls!
However FreePBX's advanced routing can be used to ensure users only have to dial 9 once for the outside line - and can still make use of the internal extensions - thus getting the best of both worlds.
1. In Trunks
set up your ZAP trunks connected to the Centrex/PABX line for your normal PSTN dial plan but coded to add "9" as if you were dialling from an extension.
For instance
9+999
9+112
9+NXXXXX
9+0.
9+1.
2. In Outbound Routes
Set up 9_outside to point to these trunks
9|.
Now your end users only have to dial "9" once and will get routed to outside line provided it matches your dialplan in trunks.
Now for the Centrex/PABX "extensions":
Choose another number as your "inter-PABX" code. I have picked "8" (as traditionally thats how many British phone systems have been set up).
Set up an outbound route to look for your Centrex/PABX extension dialling pattern after the 8 (remember that my extensions were 20-24)
8|2[01234]
Point this route to the Centrex/PABX trunks on the ZAP channel - and anything prefixed with 8 will be dialled as an "internal" extension.
Incidentally If you don't want to move your users to a longer extension digit length and the Centrex/PABX extensions are all in the same number range you can of course tweak the route to just route all numbers beginning with 2 for instance via this method, provided you don't have any FreePBX extensions in the same number range... BT Line testing and a caveat with current versions of DAHDI (as at 2009-03-15)
Many BT lines are automatically tested every night. This is to test if the cable between the Telephone Exchange and your NTE5 (the line socket) has been damaged. To do this, equipment within the exchange removes the normal dial tone and -50 volts DC (battery) from your line and substitutes it with a variety of DC and AC signals.
This appears to Zap/DAHDI hardware as a series of battery drops and line reversals
Previous versions of zaptel/DAHDI did not check the telephone line for battery voltage. This meant that the reversals picked up from the testing sequence are treated as a a "pre-ring" state (before caller ID) but time out harmlessly, so the channel was only shown unavailable for the duration of the line test (which is a sensible thing to happen).
Newer versions of DAHDI do check the line for battery. This is also sensible, as it stops Asterisk dialling into an unconnected telephone circuit.
Unfortunately, the way Asterisk is currently coded means that if the system is configured for the UK caller ID protocol, any on hook polarity reversal is treated as a precursor to the telephone ringing, when in fact (according to BT's own specifications) line reversals can randomly happen at any time.
Another piece of code inserted to make Asterisk ignore alarms at this time has inadvertently caused the combination of battery drops and reversals to be picked up as a permanent line fault.
This is under investigation at the link below, with a patch available for chan_dahdi.c
http://bugs.digium.com/view.php?id=14163
the original link from: http://www.fonicaprojects.com/wiki/index.php/Configuring_systems_for_the_UK_%28various_country_specific_quirks%29
Q68, User has to make incoming call first, then you can make outgoing calls.
certain versions of asterisk does not detect offhook correctly, there are two ways to use to make your system working as normal status. 1) plug out the fxo cable, then plug back that 2) you have to call in, then you can make outgoing calls. to solve this problem, you have to patch asterisk. this is a patch for asterisk1-6.2.2: --- org_chan_dahdi.c 2010-02-04 21:33:34.000000000 -0500 +++ new_chan_dahdi.c 2010-02-04 21:35:48.000000000 -0500 @@ -198,7 +198,7 @@
* before dialing on it. Certain FXO interfaces always think they're out of
* service with this method however.
*/
-/* #define DAHDI_CHECK_HOOKSTATE */ +#define DAHDI_CHECK_HOOKSTATE
/*! \brief Typically, how many rings before we should send Caller*ID */
1. define DEFAULT_CIDRINGS 1
@@ -10685,12 +10685,16 @@
/* When "onhook" that means no battery on the line, and thus
it is out of service..., if it's on a TDM card... If it's a channel
bank, there is no telling... */
+#ifdef DAHDI_CHECK_HOOKSTATE
if (par.rxbits > -1)
return 1;
if (par.rxisoffhook)
return 1;
else
return 0;
+#else + return 1; +#endif
} else if (par.rxisoffhook) {
ast_debug(1, "Channel %d off hook, can't use\n", p->channel);
/* Not available when the other end is off hook */
more details: https://issues.asterisk.org/file_download.php?file_id=23171&type=bug
Q69, To get the quick support, what information about the system I should send?
* Versions of kernel and Linux distribution
* Versions of asterisk and zaptel/dahdi
* Model of OpenVox products
* Debug and error information from your system and asterisk
* Zaptel(zaptel.conf and zapata.conf) or dahdi (system.conf and chan_dahdi.conf) configuration files and extension.conf if your have trouble with dialplan
* After loading the driver, run the command: demsg and send the information to us
* Output of the command: cat /proc/interrupts
* Message in asterisk console when you making a call, if you have trouble to drop/receive calls.
* Protocols you want to use(FXS,FXO,PRI..)
* Ssh account with root permission if you need us to check the system.
* Make a backup for your important files
* Describe the problem in details
* Serial numbers of OpenVox's products
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