Re: [Asterisk-Users] mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection
I also hadan issueto getmISDNworking with HFC cards without problems. Therefor I switched to zaphfc (use bristuff), this is working perfectly with HFC cards. It does everything I need including MSN support and without problems, even with multiple HFC cards. So my advice is to get rid of mISDN and to switch to zaphfc Regards Michel From: Cosmin Prund [EMAIL PROTECTED]Subject: [Asterisk-Users] mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detectionHello everyone. I've got this really annoying HFC Cologne card (orhowever I should call it - a single channel ISDN card based on the HFCchipset).It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable.Here's a sample out of CLI:P[ 1] I IND :DTMF_TONE oad:206361 dad:520101P[ 1] -- mode:TE cause:16 ocause:16 rad: cad:P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE P[ 1] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0P[ 1] -- screen:0 -- pres:0P[ 1] -- channel:1 caps:Speech pi:2 keypad:P[ 1] -- urate:0 rate:16 mode:0 user1:0P[ 1] -- pid:1 addr:50010102 l3id:30001 P[ 1] -- b_stid:10010100 layer_id:50010180P[ 1] -- bc_state:BCHAN_ACTIVATEDP[ 1] -- DTMF:*What's this all about? Is there anything I can do about it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ztcfg problem
From: Giordano Grandis [EMAIL PROTECTED]Subject: [Asterisk-Users] ztcfg problemHi all,I'm installing two HFC pci card (both in TE mode), I don't have problemwhen load module, but whrn I give ztcfg -vv, I see 6 the six channels that I configured, then my computer hang and I have to reboot it. (I'musing a VIA Epia-M 1000 with Via C3 processor)[EMAIL PROTECTED]:~# modprobe zaptel[EMAIL PROTECTED]:~# insmod /usr/src/bristuff- 0.2.0-RC8n/zaphfc/./zaphfc.o[EMAIL PROTECTED]:~# ztcfg -vv Giordano, It looks like you do a insmod of zaphfc.o while I think you have to do a insmod/modprobe of the zaphfc.ko Maybe that is causing your problem, if not, please supply us with the logging from /var/log/messages and/or other log files. Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set CLIR when using call files ?
Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now for my new issue: I want to use the CLIR for automated dial out like described on http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out There is no way (at least I was not able to find it yet) to execute the 'SetCallerPres(prohib)' before the automatic dial initiated by the call file. So my number is again visible at the called party and this is not my intention. So how should I do this? (if the keypad protocol worked it should have been simple because I could put the CLIR function in the to be dialed number but this doesnt work as I said) I tried the following: 1) Normal callfile, as described does show my number to the called party. -- 2) Callfile using local channel - does'nt work (details below) Channel: Local/[EMAIL PROTECTED]MaxRetries: 2RetryTime: 300WaitTime: 45Context: outbound_callbackserviceExtension: sPriority: 1 [localchanneldefs]exten = 50,1,SetCallerPres(prohib)exten = 50,2,Dial(Zap/4/0612345678) [outbound_callbackservice]exten = s,1,Wait(1)exten = s,2,Playback(themessage) Asterisk logging: === -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1) -- Executing SetCallerPres( Local/[EMAIL PROTECTED],2, prohib) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2 , Zap/4/0612345678) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 4/0612345678 -- Zap/4-1 is making progress passing it to Local/[EMAIL PROTECTED],2 -- Zap/4-1 is ringing -- 3) Callfile with SetCallerPres(prohib) as Application in callfile: Channel: Zap/4/0612345678 Application: SetCallerPres Data: prohib MaxRetries: 2RetryTime: 300WaitTime: 45Context: outbound_callbackserviceExtension: sPriority: 1 [outbound_callbackservice]exten = s,1,Wait(1)exten = s,2,Playback(themessage) Asterisk logging: === -- Attempting call on Zap/4/0612345678 for application SetCallerPres(prohib) (Retry 1) -- Requested transfer capability: 0x00 - SPEECHSep 2 01:37:09 NOTICE[2349]: channel.c:1865 __ast_request_and_dial: Don't know what to do with control frame 15 All attempts still show the calling number to the called party. I hope anybody has some good idea because I am lost .. Kind regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use * and # as part of number indialcommand
From: Damon Estep [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to use * and # as part of number indialcommand Good to hear you have found a temporary solution, although I think it is the permanent solution. Keypad protocol is a bandaid to fix the real problem, not a solution. The problem is that many PBXs cant set the caller ID presentation fields, so keypad protocol was derived to work around it and allow end users to modify caller ID representation from the handset 'keypads' Use the caller ID presentation flags! That is the RIGHT way to do things on ISDN. Damon Well, probably it can be a permanent solution to get the CLIR functionality but there are other functionalities which are accessible by keypad protocol like CFU *21* CFNR *61* Unless there is another dialplan command which can set these... Again , thanks for helping me out. I am happy now ;-) Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: echo system command and set the results to a new variable
From: Henry Junior [EMAIL PROTECTED]I want to issue a System cmd in my dialpan that is similar to the unix echo command stated above. *EXCEPT* that I want to pipe theresults into a *new* variable (vs a text file.)Ideally, what would happen is in my diaplan I would issue the 'echo |date' command and pipe the output to a newly created variable with the appropriate output.example:NewVariable=Mon Aug 29 18:32:45 EDT 2005 Do you know how I can do this? Assuming that you want to execute some Unix commands and feedback the output as variables to the dialplan in Asterisk this is possible with an AGI bash script. Use the 'set variable' command http://www.voip-info.org/tiki-index.php?page=set+variable You can use the example at http://yakko.cs.wmich.edu/~drclaw/asterisk/cidname/ , reuse the framework to read the standard AGI variables and rewrite it to your own code. Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc syslog flooding
From: Arik Funke arik.funke at gmx.de Hi,my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this normal?:--zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353, wanted 8 got 7), probably a buffer overrun.zaphfc: dropped audio (z1=2712, z2=2695, wanted 8 got 17, dropped 9).With Asterisk running (asterisk -vgc) but without ANY activity it prints these messages. Can anybody explain to me what the problem is? Or is there no problem?: Arik, I had the same problem until a few minutes ago. I found out that the device driver for my IDE controller was not compiled in the kernel, nor was it available as loadable module. So my IDE device was working without DMA and in 16 bit mode. That was the cause for my problems. Maybe it is the same cause for your issue. I haverecompiled the kernel with via82C (my chipset) support and now the problems are gone.The driver was found under: device drivers - ATA - via82cxxx chipset support. When I now run hdparm /dev/hda1 it show me that the device is operating in 32bit DMA mode. I just thought to let you know in case you still suffer from this issue. Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use * and # as part of number in dialcommand
Damon Estep wrote: I did not see an actual error message in your first post, what is the error message? Damon, Well, it is not a 'real' error message, asterisk logs it as a 'warning' , but for me it looks like it is linked to the problem. See my comments in the logs between [ ]. -- Executing Dial(Zap/2-1, Zap/4/*31*040268000) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1 [thus far it looks okay] -- Channel 0/1, span 2 got hangup [hmm, it seems that the channel was hangup, so it failed] Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer: Unable to forward voice [this warning indicates that asterisk was unable to forward voice, I think this is because of the *31* in the dial string, because when I leave the *31* out, the warning is not there and the connection is made without problems] -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' Thank you for your time trying to help me out! Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to use * and # as part of number indialcommand
From: Damon Estep [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to use * and # as part of number indialcommand To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? [snip] Hi Damon, Thank you for your extensive answer. The SetCallerPres(prohib) in combination with adding 'usecallingpres=yes' into zapata.conf does the job to set CLIR so I am helped out (for now). I've added this to the voip-ip wiki at SetCallerPres. I see this as a kind of workaround because I am still trying to figure out why the keypad protocol is not working as it should be. To come back on your questions: Yes it is an ISDN line, it is an ISDN card with HFC chipset, I use asterisk 1.0.7 bristuffed with zaphfc module. Switchtype: euroisdn Signalling: bri_cpe_ptmp (immediate=no, overlapdial=yes) Internally I have also ISDN HFC but in NT mode, this is where i originate the call from. In the logging this is the Zap/2-1 channel. Here is the log when I use the *31* in the dial command: -- Accepting voice call from '1001' to 's' on channel 0/2, span 1 -- Executing NoOp(Zap/2-1, ) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing Playtones(Zap/2-1, dial) in new stack -- Executing ResponseTimeout(Zap/2-1, 15) in new stack -- Set Response Timeout to 15 == CDR updated on Zap/2-1 -- Executing Dial(Zap/2-1, Zap/4/*31*) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31* -- Zap/4-1 is making progress passing it to Zap/2-1 -- Channel 0/1, span 2 got hangup Aug 29 23:18:45 WARNING[6591]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/4-1' == No one is available to answer at this time The debug log shows this (I wonder where the conferencing log is coming from because I don't use conferencing ??) Aug 29 23:18:45 DEBUG[6591]: disabled echo cancellation on channel 4 Aug 29 23:18:45 DEBUG[6591]: Set option TDD MODE, value: OFF(0) on Zap/4-1 Aug 29 23:18:45 DEBUG[6591]: Updated conferencing on 4, with 0 conference users Aug 29 23:18:45 DEBUG[6591]: Set option AUDIO MODE, value: OFF(0) on Zap/4-1 Aug 29 23:18:45 DEBUG[6591]: disabled echo cancellation on channel 4 Aug 29 23:18:45 VERBOSE[6591]: -- Hungup 'Zap/4-1' Aug 29 23:18:45 VERBOSE[6591]: == No one is available to answer at this time Aug 29 23:18:45 DEBUG[6591]: Exiting with DIALSTATUS=NOANSWER. Best regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use * and # as part of number in dial command
Hi Damon and others,
Your example is still doing what I tried already, so eventually the
dial command ends like:
Dial(zap/4/*21*)
or
Dial(zap/4/*31*)
I prefer to use Dial(zap/4/*21*thenumber)
or Dial(zap/4/*31*thenumber)
But whatever I try, the error message as in my first post shows up and
the line hangs up before the connection is made. So I assume the so
called 'keypad protocol' which is used here in the Netherlands and
possible in other countries is messing it up. I still hope there is
somebody out there who can help out.
Btw: my problem is not about how to use the Asterisk extension
(because that can be any extension where I want to put the dial
command after).
Best regards,
Michel Koenen
Damon Estep wrote:
* # are valid in a dialplan
you would start your exten = with the vertical service code *21*
then play prompt, collect digits, play prompt, dial
${exten}$(var_for_collected_digits}
BUT, unless I have missed something, You can just send *21* to the PSTN
and then follow their prompts! As long as DTMF is configured correctly
it should work. If they don't prompt you still dial the same way,
activate the vertical service code with *21*
Exten *21*,1,Dial(ZAP/CHAN/{$EXTEN})
That's all! Then just enter the rest of the digits, allowing the PSTN
switch to collect the DTMF and activate the code.
You may want to use something like this to specify which ZAP channel
your forward (or which line).
Add a line number first like this
Exten 1*21*,1,Dial(ZAP/CH1/*21*)
Notice we don't send the 1, but we do use it to pick the zap channel.
___
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[Asterisk-Users] How to use * and # as part of number in dial command
Hi all, I am struggling with the following and I can't get it work: In the Netherlands where I live it is possible to use special codes (aka vertical service codes) to set special 'behaviour' of phonecalls. So e.g. when I want to dial out with a normal phone and I dial *31*phonenumber to dial then it will turn off my numberindication (CID) at the called party. They seem to call this the 'keypad protocol' but I cannot find this term when searching through asterisk documents. My asterisk system is connected to an ISDN line with HFC card. I use zaphfc module for that. In my extensions I tried several things to dial out and use the *31* but without success. E.g.: 2000,1,Dial(Zap/4/*31*040268000) When I dial 2000 , the verbose logging shows below (Zap2-1 is my internal phone , Zap/4 is connected to outside ISDN line. -- Executing Dial(Zap/2-1, Zap/4/*31*040268000) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1 -- Channel 0/1, span 2 got hangup Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' When setting up an open link to channel and then dial the rest myself: 2001,1,Dial(zap/4,,) I get the same message as soon as I press the second * in (*31*) Is it not working because the zap channel doesn't support this ? Are there other channels who do support this? Or am I doing something wrong and do I have to use another way to get this done? Thank you in advance for your help. Best regards, Michel Koenen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automated AgentCallback logon and logoff is possible
Hi all, This is to let you know that I found out how to automate the agentcallback logon and logoff. Only thing you need, is to have the agentcode and pincode available in channelvariables. I've updated the documentation on voip-info to incorporate my findings. http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin Have fun with it! Regards, Michel Koenen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is this possible with asterisk?
Hi, I do not know of an existing application which does this out of the box. But what you want should be 'simple' to implement with AGI script(s). You need to have some programming knowledge and knowledge of AGI though. For the auto dial out there is already info on the site: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out So you have to define a context which takes care for the questions and user feedback and log them, probably AGI will help here too. Good luck Michel Koenen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 11, 2005 6:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] is this possible with asterisk? Hello Everyone! I'm wondering if the following is possible with asterisk... What i'm trying to do is find a program or a solution that can help me set appointments for a delivery company... the program should call a person asking them if the following time is suitable for a delivery... if they agree, they press one and the system logs it... if they don't agree they press two, etc... Also, another thing the system would do, would be to call the person and ask them a couple of questions and have them rate the service by pressing 1, 2, 3, etc... If anybody can point me in the right direction, it would be highly appreciated... Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to determine elapsed time of a call in progress?
Hi all, I need to be able to determining the elapsed time of a call. I tried commands like 'show channel' or 'zap show channel', this outputs a list of parameters including 'elapsed time' but for some reason this is always '0h0m0s'. Is this normal or am I looking at the wrong place or using the wrong command? My configuration: Asterisk 1.0.7 + Bristuffed The channel which I tried (and need the information for) is the channel of an agent who is handling a call via the Queue where agent is member of. This channel is running via the Zaphfc driver via HFC PCI in NT mode. Can you give me a hint to my problem? Thank you! Michel Koenen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NT1 devices with analog ports on HFC based
Yes, it is possible to connect an ISDN to analog A/B adaptor to the HFCPCI. I'm working though with the zaphfc module and the HFC card in NT mode. Then I use a (very) old PTT Moduvox 2a (only known in Netherlands I think) which is connected to the HFC card. The moduvox is a small home exchange with 1 ISDN and 2 analog connections. It provides it's own power so I don't need power on the NT1 box. Best regards, Michel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to let ZAPHFC work with and act on different incoming MSNs?
I have this working with a Teles ISA card, see config below (numbers are changed because I dont want everybody to call me;-) ) In modem.conf ZapHFC is configured in zapata.conf, not in modem.conf, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is Yes, I know but I gave the modem.conf example to show you how it is working with the Teles card. The question was how to get the same thing working with zaphfc. In the mean time I spent some more time to experiment and I found out that with zaphfc I can make use of DID to get the same results. This page gave the solution: http://voip-info.org/tiki-index.php?page=Asterisk+tips+did I just had to set immediate=no and overlapdial=yes to get it working. There is only one tiny issue left: the MSN via modem.conf is delivered as extension 402901 to the dial plan, while the MSN via zapata.conf is delivered with an extra 0 prepended so 0402901. This means that I cannot make use of the same context when using both cards. Does anybody know how to preprocess the extension before it is send to the dialplan context so that the MSN is always presented the same regardless of via which channel it is coming in? Best regards, Michel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to let ZAPHFC work with and act on different incoming MSNs?
Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have connected the ISDN line to HFC card and I use the ZAPHFC driver. I would like to tell the driver that it should jump to different places in the dial plan depending on the MSN number which was used by the caller. I have this working with a Teles ISA card, see config below (numbers are changed because I dont want everybody to call me;-) ) In modem.conf == context=incoming_teles group=0 msn=402911101 msn=402911102 device = /dev/ttyI0 device = /dev/ttyI1 == In extensions.conf: == [incoming_teles] exten = s,1,Goto(spiriimpuls,s,1) exten = 402901,1,Goto(context1,s,1) exten = 402902,1,Goto(context2,s,1) == With the ZAPHFC driver it seems that it reacts on every MSN number on that line and it always goes to the 's' extension in the specified context, it does not make use of the MSN id. How to get the same results with the ZAPHFC / zapata.conf / zaptel.conf and/or other config files, or is it not possible due to the nature of 'zap'? Thanks in advance for your contributions. Regards, Michel Koenen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gmail and the list
Same here, nothing is coming in anymore on my gmail address neither. I read your posting by going to the web version of the list. Maybe gmail is blocking mail from the list or is it really some configuration setting in the list itself ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: mISDN + chan_misdn.so + winbond issue
Hi, This is to let you all know that I have it working now. Thanks to Titus who supplied his list of a working combination ( http://amatisoft.homelinux.com/demo/index.html ) and some other tips. For archive and history purposes I will post my combination which may help others who will run into this: Packages: * kernel 2.6.9 * mISDN kernel patch from PBX4Linux (http://isdn.jolly.de/download/v3.0beta/mISDN_for_PBX4Linux_2005_03_06.tar.gz) * mISDN user from PBX4Linux (http://isdn.jolly.de/download/v3.0beta/mISDNuser_for_PBX4Linux_2005_01_28.tar.gz) * chan_misdn 0.1.0 * asterisk 1.0.7 Important factor was also to have the correct 'layermask' parameter when loading the winbond module. This had to be 0xf and not 0x1 , I am still thankful to Titus who pointed this out, at least I could not find any documentation on this parameter but it turned out to be an important one. My modprobe looks now like this: modprobe w6692pci protocol=2 layermask=0xf Best regards, Michel Koenen On 6/6/05, Michel Koenen [EMAIL PROTECTED] wrote: Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a No lower Id port:1 error. The /var/log/messages file says: MISDN free_device: entitylist not empty I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6 and against mISDNuser-CVS-2004-08-29. The /dev/mISDN node was also created. I'm loading the kernel modules this way: modprobe zaptel modprobe ztdummy modprobe mISDN_core modprobe mISDN_l1 modprobe mISDN_l2 modprobe l3udss1 modprobe mISDN_dsp modprobe w6692pci protocol=2 layermask=1 Then I start asterisk: asterisk -c -vv -dd When loading chan_misdn.so , Asterisk complains and exits after the last error line below [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISDN L2 driver version 1.20 Jun 5 20:26:02 pbx kernel: mISDN: DSS1 Rev. 1.29 Jun 5 20:26:07 pbx kernel: mISDN_dsp: Audio DSP Rev. 1.10 (debug=0x0) Jun 5 20:26:20 pbx kernel: Winbond W6692 PCI driver Rev. 1.13 Jun 5 20:26:21 pbx kernel: PCI: Found IRQ 9 for device :00:0f.0 Jun 5 20:26:21 pbx kernel: mISDN_w6692: found adapter Winbond W6692 at :00:0f.0 Jun 5 20:26:21 pbx kernel: W6692: Winbond W6692 version (0): W6692 V00 Jun 5 20:26:21 pbx kernel: w6692: IRQ 9 count 4 Jun 5 20:26:21 pbx kernel: w6692 1 cards installed Jun 5 20:26:34 pbx kernel: MISDN free_device: entitylist not empty Am I using wrong or incompatible source versions or is this a bug or am I doing something wrong? Btw the misdn.conf contains: [general] language=en immediate=no debug=0 [mycard] context=incoming ports=1,2 msns=72 Using ports=1 or ports=2 or changing msns gives the same problems.. When you have a working configuration, I am curious which source versions of needed packages you have used. Thank you in advance for your response. Best regards, Michel Koenen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mISDN + chan_misdn.so + winbond issue
David, Thank you. For some reason I'm not getting 0.1.0 build on my system, probably because I miss or have wrong version of header files or -D flags. I lost some time trying to find the ultimate combination of needed sources of the different packages but no success. Therefor I am still curious which combi of packages people use to get the misdn working. Best regards, Michel On 6/7/05, David Phelan [EMAIL PROTECTED] wrote: Update to at least chan_misdn-0.1.0 .. I am using snapshot from 11.05.05 without too many issues. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michel Koenen Sent: Monday, 6 June 2005 6:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mISDN + chan_misdn.so + winbond issue Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a No lower Id port:1 error. The /var/log/messages file says: MISDN free_device: entitylist not empty I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6 and against mISDNuser-CVS-2004-08-29. The /dev/mISDN node was also created. I'm loading the kernel modules this way: modprobe zaptel modprobe ztdummy modprobe mISDN_core modprobe mISDN_l1 modprobe mISDN_l2 modprobe l3udss1 modprobe mISDN_dsp modprobe w6692pci protocol=2 layermask=1 Then I start asterisk: asterisk -c -vv -dd When loading chan_misdn.so , Asterisk complains and exits after the last error line below [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISDN L2 driver version 1.20 Jun 5 20:26:02 pbx kernel: mISDN: DSS1 Rev. 1.29 Jun 5 20:26:07 pbx kernel: mISDN_dsp: Audio DSP Rev. 1.10 (debug=0x0) Jun 5 20:26:20 pbx kernel: Winbond W6692 PCI driver Rev. 1.13 Jun 5 20:26:21 pbx kernel: PCI: Found IRQ 9 for device :00:0f.0 Jun 5 20:26:21 pbx kernel: mISDN_w6692: found adapter Winbond W6692 at :00:0f.0 Jun 5 20:26:21 pbx kernel: W6692: Winbond W6692 version (0): W6692 V00 Jun 5 20:26:21 pbx kernel: w6692: IRQ 9 count 4 Jun 5 20:26:21 pbx kernel: w6692 1 cards installed Jun 5 20:26:34 pbx kernel: MISDN free_device: entitylist not empty Am I using wrong or incompatible source versions or is this a bug or am I doing something wrong? Btw the misdn.conf contains: [general] language=en immediate=no debug=0 [mycard] context=incoming ports=1,2 msns=72 Using ports=1 or ports=2 or changing msns gives the same problems.. When you have a working configuration, I am curious which source versions of needed packages you have used. Thank you in advance for your response. Best regards, Michel Koenen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 4/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 4/06/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN + chan_misdn.so + winbond issue
Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a No lower Id port:1 error. The /var/log/messages file says: MISDN free_device: entitylist not empty I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6 and against mISDNuser-CVS-2004-08-29. The /dev/mISDN node was also created. I'm loading the kernel modules this way: modprobe zaptel modprobe ztdummy modprobe mISDN_core modprobe mISDN_l1 modprobe mISDN_l2 modprobe l3udss1 modprobe mISDN_dsp modprobe w6692pci protocol=2 layermask=1 Then I start asterisk: asterisk -c -vv -dd When loading chan_misdn.so , Asterisk complains and exits after the last error line below [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISDN L2 driver version 1.20 Jun 5 20:26:02 pbx kernel: mISDN: DSS1 Rev. 1.29 Jun 5 20:26:07 pbx kernel: mISDN_dsp: Audio DSP Rev. 1.10 (debug=0x0) Jun 5 20:26:20 pbx kernel: Winbond W6692 PCI driver Rev. 1.13 Jun 5 20:26:21 pbx kernel: PCI: Found IRQ 9 for device :00:0f.0 Jun 5 20:26:21 pbx kernel: mISDN_w6692: found adapter Winbond W6692 at :00:0f.0 Jun 5 20:26:21 pbx kernel: W6692: Winbond W6692 version (0): W6692 V00 Jun 5 20:26:21 pbx kernel: w6692: IRQ 9 count 4 Jun 5 20:26:21 pbx kernel: w6692 1 cards installed Jun 5 20:26:34 pbx kernel: MISDN free_device: entitylist not empty Am I using wrong or incompatible source versions or is this a bug or am I doing something wrong? Btw the misdn.conf contains: [general] language=en immediate=no debug=0 [mycard] context=incoming ports=1,2 msns=72 Using ports=1 or ports=2 or changing msns gives the same problems.. When you have a working configuration, I am curious which source versions of needed packages you have used. Thank you in advance for your response. Best regards, Michel Koenen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
