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=== 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)=== 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=== 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?=== 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=== 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/>
===Q7, Call conversation suddenly dropped=== 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?=== 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?=== 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?=== 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?=== 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/
===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.<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
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?=== 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.
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謝謝分享這麼詳盡的資料,收下了。
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===
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)===
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===
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?===
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===
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===
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===
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?===
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?===
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?===
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?===
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===
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===
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?===
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?===
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?===
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/>
===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===
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/>
多交流一些,确实能少走一些弯路。
其实我觉得真正能帮助其他人也是一个对社区的贡献,至少能够让一些asterisk 人少走一些弯路。准备改进一下工作方法,看看能否奏效。
樓主的用心良苦,小弟可以體會,也希望這些挫折不會讓樓主失去了那份最初的分享技術的熱情才好。
真正沈溺於技術研究的那些玩家,是一點都不會在乎高手,低手的分別的。很多事情若換個角度思考,就會有不一樣的心情,直到最後才又發現,這些困擾皆是 "庸人自擾之" 。
hi:
如果按照你自己现在的状况,我建议你最好先 玩 asterisk + centos+sip 软电话。你可以先试试 内部分机之间的互打。你虚拟机安装就可以了。不需要什么其他的硬件资源。当然你慢慢以后可以买 一张模拟考去尝试设置呼入和呼出,设置 ivr.....
刚开始的时候你必须了解 asterisk 的基本的设置,例如 sip 分机如何设置,extension.conf 是怎么回事。慢慢在学习其他的功能。至于 elastix, trixbox, easyasterisk, broker, asterisknow, pbx in flash, 这些就是把asterisk和 linux 搞在了一起,然后加了一个 freepbx 或者类似 freepbx 的php 界面去控制 extension.conf, sip.conf 还有问题的 etc/asterisk 下面相关的文件。 如果你学习leastix, 你最好搞清楚 elastix 之类系统的设置流程,要不然会晕。最后没有兴趣了。我还是建议你找一些好的网站,慢慢一步一步的去学习,从 asterisk 里面学,知道这些配置文件的功能,然后学习dialplan. 搞开始都是这样的,肯定会模糊一段时间,慢慢就知道了。这些网站估计你用看过:
asteriskguru.com, voip-info.org, 里面的东西相当不错的。 希望下一个 asterisk 人才从这里诞生!
zhulizhong
zhulizhong 你好 我是一個asterisk 的新手.... 我對linux 都有一些基本的認識 可是對asterisk 搞了幾個月都...沒有很什麼進步. 上網問人... 沒人回答我 可是是我的問題...太沒深度...所以他們不管我 我現在己玩asterisk一個簡單的version 就是
Elastix. 都有一些問題解決不了 你覺得我應該用elastix 還是努力玩asterisk 謝謝
我目前观察到的,现在的国内市场就是这样的一个状态:
1 ) 我们必须先培训客户,给他们讲如何用 asterisk, 然后有什么应用。
2)由于第一个问题,导致了客户问很多和我们产品不相关的问题,回答了他们高兴,不回答他们就说你服务不好。
3) 我们公司对国内客户来说,基本成了一个asterisk 的咨询机构了,普及asterisk 成了我们的一个任务。40% 的时间重复就像www.cnasterisk.com 的版主 leeelton 说的,浪费时间。这里面有个问题就是可能一些搞应用的厂商,你们遇到的客户是一个一个来到或者是项目一个一个的做,时间上面冲突的机会不多。我们的客户是每天来自不同的国家不同的问题。如果客户的问题提到很不清楚,或者没有给相关的信息,很让人郁闷。搞的一天都没有心情做事。
4) 现在好像是一张白纸,那其实还有一些机会,大家都会了,我们就没有机会了。赚钱的机会就是0 了。但是我们必须有一个好的规范,在这个论坛上面看到的就比较好,大家发帖子就发到该发的地方,不要乱发广告。现在发现很多论坛都没有规范,人多了,什么人都来折腾,什么垃圾都往上发,那就麻烦了。
5) 技术服务范围的不确定性。这里涉及到了客户和厂商如何平衡的问题。搞不明白如何管理这里。
6)技术资源的整合问题。各种设备的数字,功能的设置。其实难度还是很大的。有些时候没有那个环境,其实很难做到完整。这也是为什么asterisk 的资料什么地方都有,但是都不是很完整。只能是逐渐完善而已。这里又回到了我说的中国人的心态问题,其实可能就那么几个人写的,但是大部分都是转来的,把别人的名字换掉,发在自己的博客里面,貌似水平挺高。
7) 很多asterisk 的人,不知道报着什么心态,互相诋毁对方,或者把别人说的一无是处。其实那样只能说明自己心虚,自己没有自信。论坛用户之间可以评价 什么软件好什么不好,但是厂商绝对不能说自己的观点,认为自己好。幼稚! 自己必须保持中立。市场用户都有认知的,大家都不傻。君子以艺治人,以德治人。就像我们领导人说的“以德治国”, 那才是王道。但是德好像在中国成立稀罕的东西。 唉,说归说,真正做起来很难啊。
zhulizhong
[ 本帖最后由 zhulizhong 于 2009-6-13 11:52 编辑 ]