Re: [asterisk-users] Alarm events + asterisk dies
Hello, Thank you for the advice. I am sorry but I could not locate the problem in the forum. Do you remember anything more specific about it? And was it on asterisk-users? Do you remember year and month when it was seen? Thanks a lot, Roberts On Tue, 2008-10-14 at 05:20 -0400, broadband Voice wrote: You need to download a patch for zaptel, thats why your server is crushing. Search through the forum, there is a known problem or reverse to a version of Asterisk that is compatible with you zaptel. On Tue, Oct 14, 2008 at 2:19 AM, Roberts Klotins [EMAIL PROTECTED] wrote: Hello there, With extended logging options the events just before Asterisk dying look like this: [Oct 14 00:52:45] VERBOSE[2496] logger.c: == Starting post polarity CID detection on channel 1 [Oct 14 00:52:45] DEBUG[2496] dsp.c: dsp busy pattern set to 500,500 [Oct 14 00:52:45] VERBOSE[3188] logger.c: -- Starting simple switch on 'Zap/1-1' [Oct 14 00:52:47] NOTICE[3188] chan_zap.c: Got event 4 (Alarm)... [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Ignoring Polarity switch to IDLE on channel 1, state 9 [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -123711628 [Oct 14 00:52:48] NOTICE[3188] chan_zap.c: Alarm cleared on channel 1 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 9 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignoring Polarity switch to IDLE on channel 1, state 9 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -123710540 And that is the last event. Channel 1 is FXO port where my BT line is plugged in. Can anyone suggest if it seems this may be a card fault, or have I misconfigured something? I would really appreciate your help, I cannot afford to have asterisk die randomly. Roberts On Mon, 2008-10-06 at 08:26 +0100, Roberts Klotins wrote: Hi All, I am getting these events in asterisk message log: NOTICE[16647] chan_zap.c: Got event 4 (Alarm)... NOTICE[16647] chan_zap.c: Alarm cleared on channel 1 after that asterisk exits silently until I restart it. Sometimes zapata drivers also get in a state where I need to physically restart the machine. Does anyone have any suggestions how to troubleshoot these alarm events? Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alarm events + asterisk dies
Hello there, With extended logging options the events just before Asterisk dying look like this: [Oct 14 00:52:45] VERBOSE[2496] logger.c: == Starting post polarity CID detection on channel 1 [Oct 14 00:52:45] DEBUG[2496] dsp.c: dsp busy pattern set to 500,500 [Oct 14 00:52:45] VERBOSE[3188] logger.c: -- Starting simple switch on 'Zap/1-1' [Oct 14 00:52:47] NOTICE[3188] chan_zap.c: Got event 4 (Alarm)... [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Ignoring Polarity switch to IDLE on channel 1, state 9 [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -123711628 [Oct 14 00:52:48] NOTICE[3188] chan_zap.c: Alarm cleared on channel 1 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 9 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignoring Polarity switch to IDLE on channel 1, state 9 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -123710540 And that is the last event. Channel 1 is FXO port where my BT line is plugged in. Can anyone suggest if it seems this may be a card fault, or have I misconfigured something? I would really appreciate your help, I cannot afford to have asterisk die randomly. Roberts On Mon, 2008-10-06 at 08:26 +0100, Roberts Klotins wrote: Hi All, I am getting these events in asterisk message log: NOTICE[16647] chan_zap.c: Got event 4 (Alarm)... NOTICE[16647] chan_zap.c: Alarm cleared on channel 1 after that asterisk exits silently until I restart it. Sometimes zapata drivers also get in a state where I need to physically restart the machine. Does anyone have any suggestions how to troubleshoot these alarm events? Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alarm events + asterisk dies
Thanks for the reply. asterisk-1.4.21.2 zaptel-1.4.12.1 I have posted a more detailed error log in this thread. Roberts On Tue, 2008-10-14 at 11:54 +0200, Tzafrir Cohen wrote: On Mon, Oct 06, 2008 at 08:26:30AM +0100, Roberts Klotins wrote: Hi All, I am getting these events in asterisk message log: NOTICE[16647] chan_zap.c: Got event 4 (Alarm)... NOTICE[16647] chan_zap.c: Alarm cleared on channel 1 after that asterisk exits silently until I restart it. Sometimes zapata drivers also get in a state where I need to physically restart the machine. Does anyone have any suggestions how to troubleshoot these alarm events? What versions of asterisk and zaptel? zaptel should not be sending and clearing an alarm in such a case. I think it was solved in recent versions of zaptel (raising the threashold for detecting an alarm). OTOH, Asterisk should not crash just because. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interpreting Asterisk Logs
On Thu, 2008-10-09 at 12:51 +0800, Darren Murphy wrote: Hi, Can anybody point me to an online resource that will assist with interpreting Asterisk log files? I note that a similar question was asked in this forum some time ago (http://lists.digium.com/pipermail/asterisk-users/2007-June/189793.html), which doesn't appear to have received any responses. On that occasion, the OP was seeking a log parser - I'm looking for more of a general reference guide. I'm quite new to Asterisk, and VOIP in general, and I'm struggling to understand what many of logged messages mean. The current approach I am taking is to google for specific messages (or parts thereof) - and this has been somewhat fruitful, if not quite tedious. It would be nice to have a reference guide that lists the most common log messages, and what they mean. Does such a guide exist? thanks, Darren Hi Darren, I also would find such a guide very helpful. Probably something is documented in source code. It would be interesting to know whether the kind people who are working on asterisk documentation project have had thoughts on this aspect. Regards, Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alarm events + asterisk dies
Hi All, I am getting these events in asterisk message log: NOTICE[16647] chan_zap.c: Got event 4 (Alarm)... NOTICE[16647] chan_zap.c: Alarm cleared on channel 1 after that asterisk exits silently until I restart it. Sometimes zapata drivers also get in a state where I need to physically restart the machine. Does anyone have any suggestions how to troubleshoot these alarm events? Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cellroute setup with asterisk
OK, here is how it is working so far: Cellroute has a 3G sim card in it. Its Phone port is connected to TDM400P FXO port, just next to my incoming BT line. Calling out works fine - just as with the BT line. On incoming calls there seems to be a problem with caller ID chan_zap.c:4155 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. Also does not get me to voicemail with error messages as follows: -- Zap/4-1 is ringing -- Nobody picked up in 15000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:4] Goto(Zap/2-1, s-NOANSWER|1) in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [EMAIL PROTECTED]:1] VoiceMail(Zap/2-1, 10| u) in new stack -- Zap/2-1 Playing 'vm-theperson' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Zap/2-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' However when I set up the incoming call to go directly to voicemail everything seems to work alright. Next things to do will be to create a voice menu and see how well DTMF tones are recognised. Then I plan to add Cellroute to LAN (for sending SMS) and to serial port (so that the default IP address can be changed and perhaps other useful things done as well). Robert On Wed, 2008-09-17 at 09:04 +0100, Roberts Klotins wrote: Hi there! Sorry, I should have started this as a separate thread. Here we go: I wonder if anyone has set up Cellroute or Cellroute 3G mobile network gateway (see http://www.gsmsave.com/acatalog/CellRoute-3G.pdf ) with asterisk. I am about to do that soon, therefore any experience would be highly appreciated. I understand that one could connect the PSTN port on it to a FXO port on a TDM400P card and that probably could take care of calling. I wonder how then is it possible to deal with SMS? Best wishes, Robert P.S. And of course I will be posting followups to inform how I am getting along with the setup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cellroute setup with asterisk
Hi there! Sorry, I should have started this as a separate thread. Here we go: I wonder if anyone has set up Cellroute or Cellroute 3G mobile network gateway (see http://www.gsmsave.com/acatalog/CellRoute-3G.pdf ) with asterisk. I am about to do that soon, therefore any experience would be highly appreciated. I understand that one could connect the PSTN port on it to a FXO port on a TDM400P card and that probably could take care of calling. I wonder how then is it possible to deal with SMS? Best wishes, Robert P.S. And of course I will be posting followups to inform how I am getting along with the setup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK call initiating party hangup control on analoghome lines
Thanks Don, I have looked at that page and the described problem (when asterisk does not detect remote (calling) party hangup) is exactly the opposite to my case. In my case I learnt that for BT residential analog lines the remote caller is entitled to hold the line open indefinitely, regardless whether I hang up or not. So this seems to be a feature of BT rather than a problem with asterisk. I do not know whether there are many countries where telco setup is similar. But looking at the page again I discovered a pointer to this very interesting document: Voice Network Signalling and Control http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml So cheers, Robert On Mon, 2008-09-15 at 07:44 -0500, Don Kelly wrote: I don't know what options BT may provide, but check out calling party control on the Asterisk wiki and see if there's something you can do in your configuration to make it all better: http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 15, 2008 6:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] UK call initiating party hangup control on analoghome lines I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how long the line will stay open once it is connected. Example: When asterisk box receives a call, it answers, perhaps takes a voice message and issues hangup(). If the calling party does not hang up - the line remains connected - i.e. the caller effectively controls the line. Of course the caller pays for it to BT, therefore it is not a problem for BT. On asterisk irc channel I received a reply along the lines the caller controls the line, it has always been like that in telephony and also a reply that in Germany it is different. My experience in Latvia is also different - if any party hangs up, the other party hears busy signal and line is disconnected. Kind of makes more sense if you are used to it. I am posting to provide information for people who might be using landlines in UK and become similarly confused. There is very little information about this, I couldn't find it on the web, people on asterisk irc also were at a loss, BT phone support themselves thought this was a fault. It took an engineer to be able to explain that this was a UK PSTN feature by design. This is not asterisk problem - for example asterisk in my case detects remote hangups alright (which usually has been the problem with UK lines and about which there is quite a bit of info on www), and it can hangup properly if it has initiated a call. So a person calling from say Vodafone mobile would have to be careful to actually make sure he/she presses the red button after speaking to my answering machine on Asterisk. Or any answering machine for that matter, since the problem of course is reproducible with any phone. Best wishes, Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK call initiating party hangup control on analoghome lines
Hi Gordon, The test would be call your home BT line, hang up the home end and THEN wait for how long time the line will stay open listening to the handset that initiated the call. If it is the caller (call initiator) who hangs up, line is released very quickly. But perhaps thats what you are doing, since you say that the line is cleared inside 10 minutes (I have never waited that long). If I call from T-mobile - line is cleared in 30 sec, if I call from Vodafone - it stays open over 2 minutes; then I hang up Vodafone. If the call is made from a landline - I also have waited over 2 minutes but not 10. I am not sure what the difference is when you order the line and tell about PABX, however it seems to me that if I now asked to set up my existing line for PABX, they might say that this is not a business line, therefore they can't help, mightn't they? Robert On Tue, 2008-09-16 at 21:07 +0100, Gordon Henderson wrote: On Tue, 16 Sep 2008, Roberts Klotins wrote: Thanks Don, I have looked at that page and the described problem (when asterisk does not detect remote (calling) party hangup) is exactly the opposite to my case. In my case I learnt that for BT residential analog lines the remote caller is entitled to hold the line open indefinitely, regardless whether I hang up or not. So this seems to be a feature of BT rather than a problem with asterisk. I do not know whether there are many countries where telco setup is similar. That might be worth a posting to usenet:uk.telecom ... I've a funny feeling this is exchange dependant and it's not the same action for all exchanges. I can call my home BT line, answer it, then hangup and the call is cleared inside 10 minutes (usually 2-3) then I can make an outgoing call. I'm told that you need to tell BT that you have a PBX on the end of the line, but I've never done that for any analogue installation I've done. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK call initiating party hangup control on analoghome lines
Hi Don, All that you have posted is true however it is so for cases when asterisk does not detect the caller from outside has hang up - and continues to record VM thinking the caller is still speaking. My problem however is not in detecting remote hangups, but asterisk hanging up and remote end (the caller) keeping the line open by not hanging up. Robert On Tue, 2008-09-16 at 15:47 -0500, Don Kelly wrote: The wiki had this about UK POTS: In the UK, if you are using BT and are having problems with Asterisk not detecting hangups when using a TDM400P, contact BT and ask what the current Disconnect Clear Time setting is for your phone line. Mine was set to 100 and increasing it to 800 fixed the issue. Disconnect Clear Time (DCT) is BT's name for CPC. You will need to speak to the BT Faults service--they can check to see what your DCT/CPC is set to and reset it instantly to another value. BT have two analogue line specifications: single line (BT SIN 351), and multiline (BT SIN 352), which explain the different forward disconnect parameters (http://www.sinet.bt.com/). These specifications state that for single line (like most of us have), the disconnect time is between 90ms and 130ms, for multiline it is 800ms. My impression is that the rest of the World (North America included) has a disconnect time in excess of 500ms, this explains why the default behaviour of zaptel (Digium and Sangoma cards alike) don't play well with single line BT phone lines for disconnect supervision. When ordering lines from BT - especially feature lines or business lines ask the sales person to set lines to PABX working. And if none of this works, it looks like the maxsilence parameter could end voice mail recording and allow you to terminate the call. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Tuesday, September 16, 2008 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] UK call initiating party hangup control on analoghome lines On Tue, 16 Sep 2008, Roberts Klotins wrote: Thanks Don, I have looked at that page and the described problem (when asterisk does not detect remote (calling) party hangup) is exactly the opposite to my case. In my case I learnt that for BT residential analog lines the remote caller is entitled to hold the line open indefinitely, regardless whether I hang up or not. So this seems to be a feature of BT rather than a problem with asterisk. I do not know whether there are many countries where telco setup is similar. That might be worth a posting to usenet:uk.telecom ... I've a funny feeling this is exchange dependant and it's not the same action for all exchanges. I can call my home BT line, answer it, then hangup and the call is cleared inside 10 minutes (usually 2-3) then I can make an outgoing call. I'm told that you need to tell BT that you have a PBX on the end of the line, but I've never done that for any analogue installation I've done. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cellroute setup with asterisk
Hi there! I wonder if anyone has set up Cellroute or Cellroute 3G mobile network gateway (see http://www.gsmsave.com/acatalog/CellRoute-3G.pdf ) with asterisk. I am about to do that soon, therefore any experience would be highly appreciated. I understand that one could connect the PSTN port on it to a FXO port on a TDM400P card and that probably could take care of calling. I wonder how then is it possible to deal with SMS? Best wishes, Robert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
