asterisk 模拟卡常见问题以及解决办法 第一部分

zhulizhong

This troubleshooting is conducted to give users a guideline to fix
problems. here, most of problems are list out, if user follow  that exactly, most of the problems should be solved.
===Q1, You can not compile zaptel and asterisk=== // 不能编译asterisk 和 zaptel
please make sure that:<br/>
1) You have installed all necessary packages and kernel source.<br/>
2) Make sure the version of kernel source is exactly same with the version of the kernel.<br/>
please check the few links:<br/>
http://wiki.openvox.cn/index.php/A1200P<br/>
http://wiki.openvox.cn/index.php/A400P<br/>
http://www.asteriskguru.com/tutorials/<br/>
3) make sure that you do not miss any packages or files in asterisk or zaptel.<br/>
4) make sure your system can access www.asterisk.org.<br/>

===Q2, ZT_SPANCONFIG failed on span 1: Invalid argument (22)=== // ztcfg 启动  的问题
please check:<br/>
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.<br/>
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:<br/>
http://www.openvox.cn/kb/entry/2/<br/>
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.<br/>
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.<br/>
5) if you still can not boot it up, you have to recompile zaptel or dahdi again.<br/>

===Q3, You can not make calls from asterisk=== // 不能呼入或者呼出
there are few reasons why you can not make calls:<br/>
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<br/>
2) your wctdm or opvxa1200 does not boot up(leds are off).<br/>
3) leds are up and card driver has boot up properly, but the zapata.conf is<br/>
, so asterisk does not boot up properly,<br/>
please check by run: zap show channels<br/>
if is empty or no such command, you should check your zapata.conf<br/>
4) You maybe recompile your zaptel and asterisk again.<br/>

===Q4, How do you adjust the volume of voice for analog cards?=== // 然后调整音量
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:<br/>
http://linux.die.net/man/8/ztmonitor

===Q5, You can not hangup calls=== // 不能挂断电话
To resolve the problem, please check:<br/>
1) set timezone and defaulzone to your country, set country=your country in indication.conf and run: modprobe wctdm/opvxa1200 opermode=YOUR country<br/>
2) open busydetect=yes and busycount=4<br/>
3) ask your provider to open the "disconnect supervision" service
check for more details,<br/>
please go here:<br/>
http://www.asteriskguru.com/tuto ... _tdm_voicemail.html<br/>

===Q6, You can not get the callerid=== // 没有来电显示的问题
If you have a problem with callerid, please check with this link:<br/>
http://bbs.openvox.cn/viewthread.php?tid=831&extra=page%3D1

===Q7, Call conversation suddenly dropped=== // 掉线的问题
please refer this reference from digium:<br/>
Dropped Calls on TDM<br/>
If you are having dropped calls on a TDM400P card or an X100P card there are several things that might cause this.<br/>
1)BusyDetect<br/>
2)CallProgress<br/>
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.<br/>
link:http://kb.digium.com/entry/71/

===Q8, How can you set the analog card for your country?=== // 设置国家制式
To set the pbx with your country support, you must:<br/>
1) set timezone and defaultzone to your country in zaptel.conf or system.conf of dahdi<br/>
2) set the country=your country in indication.conf<br/>
3) modprobe wctdm or opvxa1200 opermode=YOUR country with capital letter.<br/>
4) after load the drivers, run dmesg command to check the mode.<br/>

===Q9, How can you open the debug for asterisk?=== // 打开debug 文件
1) You can edit the  file logger.conf under /etc/asterisk,<br/>
enable the debug or error, those message will be stored under<br/>
/var/log/asterisk<br>
2) you also can start your asterisk in this way:<br/>
asterisk -vvvvvvvvgc -d

===Q10, How can i check the IRQ of analog cards?=== // 中断号的检查
please run the command:<br/>
cat /proc/interrupts<br/>
you should see the IRQs, Make sure the card has OWN IRQ, Do NOT share with other devices.<br/>
more details, please check from here:<br/>
http://www.voip-info.org/wiki/vi ... bus+Troubleshooting

===Q11, Where is the opvxa1200 drivers user manuals for dahdi and zaptel?=== // 启动相关文件下载
Under the download, you can see that there are three subdirectories:<br/>
First one is driver, you can get the individual opvxa1200 driver.<br/>
Second is a zaptel with opvxa1200, you can choose a proper version for you.<br/>
Third one is for dahdi, if you want to try dahdi, you can download whole packages.<br/>
link: http://www.openvox.cn/download/

===Q12, Sound Quality Problems with Analog cards=== // 声音的解决办法
please refer this link:<br/>
http://www.asteriskguru.com/tuto ... p_te405p_noise.html

===Q13, How can you compile asterisk with dahdi for wctdm and opvxa1200=== // 编译 dahdi
please refer these links:<br/>
http://bbs.openvox.cn/viewthread.php?tid=574&extra=page%3D3<br/>
http://bbs.openvox.cn/viewthread.php?tid=587&extra=page%3D1<br/>
http://www.openvox.cn/download/<br/>
http://www.voip-info.org/wiki/view/DAHDI<br/>
http://www.russellbryant.net/blog/category/dahdi/<br/>
http://blog.paulsnet.org/?p=44<br/>
http://docs.tzafrir.org.il/dahdi-tools/?C=S%3BO=A<br/>

===Q14, I am hearing an echo. What can I do to fix this?=== // 回事抑制的问题
please refer these links:<br/>
http://kb.digium.com/entry/1/<br/>
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation<br/>

===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?===// 指示灯的意思
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.<br/>

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:<br/>
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.<br/>

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. <br/>

===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.<br/>

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.<br/>

Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it.<br/>

===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.'<br/>

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.<br/>

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:

# 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. <br/>
link:http://kb.digium.com/entry/61/

===Q21, Why am I having DTMF detection problems?=== // dtmf 的问题
Zaptel DTMF Detection Problems<br/>
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.<br/>

SIP DTMF Detection Problems<br/>
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.<br/>

===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<br/>
www.trixobx.org<br/>
。。。。很多
===Q26, How can you install asterisk with Debian Ubutun===
http://www.debianhelp.co.uk/asterisk.htm<br/>
http://www.itinfusion.ca/asteris ... isk-on-debian-etch/<br/>
http://www.voip-info.org/tiki-in ... terisk+Linux+Debian<br/>
http://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian<br/>
http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu<br/>
http://ubuntuforums.org/showthread.php?t=136785<br/>

===Q27, How can you install asterisk with Fedora?===
http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora<br/>
http://www.asteriskguru.com/<br/>

===Q28, How can you install asterisk with SuSe?===
http://www.asteriskguru.com/tuto ... mpilation_suse.html<br/>
http://voip-manager.net/installation-linux-asterisk.php<br/>

===Q29, install asterisk with Free BSD===
http://www.voip-info.org/wiki/view/Asterisk+FreeBSD<br/>
http://www.voip-info.org/wiki/view/FreeBSD+zaptel<br/>

===Q30, List of Asterisk OS Platforms=== 支持 asterisk 的linux unix 的发布版
http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms<br/>

===Q31, Centos with asterisk===
http://www.voip-info.org/wiki/vi ... +1.6.x+installation<br/>
http://www.voip-info.org/wiki/vi ... +1.4.x+installation<br/>
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos<br/>
以上是第一部分，里面有很多是其他网站的链接，只是问题的汇总已经解决办法。以后会陆续出来其他的部分。
zhulizhong