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VoIP88

james.zhu 发表于 2009-6-26 21:42 | 显示全部楼层 |阅读模式
本帖最后由 james.zhu 于 2009-6-27 16:37 编辑

===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
7) Make sure the physical connection is well established 8) 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 digital 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/vi ... bus+Troubleshooting

===Q7, Sound Quality Problems with Digital cards===

please refer this link:
http://www.asteriskguru.com/tuto ... p_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/


===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 "CI 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/asteris ... isk-on-debian-etch/
http://www.voip-info.org/tiki-in ... terisk+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/tuto ... mpilation_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/vi ... +1.6.x+installation
http://www.voip-info.org/wiki/vi ... +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) 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

===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/tuto ... p_te405p_noise.html
2) zttool
http://www.voip-info.org/wiki/view/Asterisk+zttool
3) ztmonitor
http://www.voip-info.org/wiki/vi ... ata+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 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/viewtop ... 37576c7aa92518fe48b

===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/pipermai ... 07-June/189259.html

===Q35, Bug#439814: zaptel-source: oslec_echo_can_identify undefined symbol===

please refer this:
http://lists.alioth.debian.org/p ... -August/009225.html

===Q36, How to install Octasic SoftEcho===

please refer these links:
http://www.openvox.cn/download/u ... n/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/vi ... dc698e89467c3d49a86

===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->ort 1 connected to TELCO // port 1 MUST be Normal(slave) clock mode
zaptel->ort 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.
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



===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

点评

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爱斯瑞科 发表于 2012-7-16 15:30 | 显示全部楼层
收藏起来,
entoKxlW 发表于 2012-9-6 18:55 | 显示全部楼层
强烈感谢楼主








irrwu 发表于 2013-1-9 23:21 | 显示全部楼层
真是好人啊~~








uuwang038 发表于 2017-9-5 03:39 | 显示全部楼层

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linedd963 发表于 2017-9-28 02:31 | 显示全部楼层
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