Re: [asterisk-users] SIP Debugging Information..
Michael L. Young wrote: If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Prior to that, Asterisk retransmits the OK to Bandwidth.com's INVITE twice. It doesn't look like Bandwidth.com receives any of them, because they never respond with an ACK. Since, from Bandwidth.com's perspective, the call is never setup, they terminate it with a BYE. It could just be a NAT issue, but there are two things I really don't understand about the SIP dialog: 1) It starts with an ACK from Bandwidth.com. Is it possible that the debugging output is missing the beginning of the dialog? 2) Every timestamp is Nov 23 15:43:13. I don't think the SIP session timers on either end should be expiring quickly enough for this to happen. Do other calls originating from Bandwidth.com work properly? If so, comparing the SIP from a working call to a failed call may be revealing. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Debugging Information..
I did a little googling, but didn't seem to find anything specific to answer the question. I am trying to debug some calls on an Asterisk system (AsteriskNow) that are dropping, and when the general logs didn't nail anything I turned on SIP Debugging on the trunk to the provider. Basically the complaint is that when some call in, regardless of if the call is answered, or if Vmail answers it, it drops the calls in a matter of seconds. The strange thing is, that the system processes many hundreds of calls daily, but only a couple specific incoming callers are seeing the drops. I would have thought a NAT issue, but why does this only affect a specific group of incoming callers, the rest go about their business just fine. I think thinking bandwidth.com is mucking something up, but again I have no specific proof one way or another, so why the debugging. When one of the problem callers is dropped, in the SIP debugging I see: chan_sip.c: Scheduling destruction of SIP dialog '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) Is this the remote end (ie bandwidth.com) dropping the call, or is the local Asterisk server dropping the call? For any that care to look at all the gory details, here is a chunk of the debugging output: [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- ACK sip:4104159233@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKebf3.453cc5a5.2 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKebf3.315d4e14.3 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f5f0b80116aa493 From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11346 ACK Max-Forwards: 68 Content-Length: 0 - [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- (12 headers 0 lines) --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- INVITE sip:4104159233@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKfbf3.9d9b1065.0 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKfbf3.c159a6a.0 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f601a4d116aa493 From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11347 INVITE Max-Forwards: 68 Contact: sip:+12159470824@192.168.37.72:5060 Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 22153 5058 IN IP4 192.168.37.72 s=SIP Media Capabilities c=IN IP4 67.231.8.102 t=0 0 m=audio 6576 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- (15 headers 11 lines) --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Sending to 216.82.224.202:5060 (NAT) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found RTP audio format 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found RTP audio format 101 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Peer audio RTP is at port 67.231.8.102:6576 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- Transmitting (NAT) to 216.82.224.202:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKfbf3.9d9b1065.0;received=216.82.224.202;rport=5 060 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKfbf3.c159a6a.0 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f601a4d116aa493 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11347 INVITE Server: FPBX-2.9.0(1.8.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: sip:4104159233@10.98.4.36:5060 Content-Length: 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Audio is at 11444 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: ---
Re: [asterisk-users] SIP Debugging Information..
- Original Message - From: Howard Leadmon how...@leadmon.net To: asterisk-users@lists.digium.com Sent: Saturday, November 24, 2012 3:19:10 PM Subject: [asterisk-users] SIP Debugging Information.. I did a little googling, but didn't seem to find anything specific to answer the question. I am trying to debug some calls on an Asterisk system (AsteriskNow) that are dropping, and when the general logs didn't nail anything I turned on SIP Debugging on the trunk to the provider. Basically the complaint is that when some call in, regardless of if the call is answered, or if Vmail answers it, it drops the calls in a matter of seconds. The strange thing is, that the system processes many hundreds of calls daily, but only a couple specific incoming callers are seeing the drops. I would have thought a NAT issue, but why does this only affect a specific group of incoming callers, the rest go about their business just fine. I think thinking bandwidth.com is mucking something up, but again I have no specific proof one way or another, so why the debugging. When one of the problem callers is dropped, in the SIP debugging I see: chan_sip.c: Scheduling destruction of SIP dialog '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) Is this the remote end (ie bandwidth.com) dropping the call, or is the local Asterisk server dropping the call? [snip] --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- BYE sip:4104159270@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0b66d829 Record-Route: sip:67.231.4.93;lr=on;ftag=gK0b66d829 Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKe902.53bde7e.0 Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bKe902.32697e93.0 Via: SIP/2.0/UDP 192.168.27.72:5060;branch=z9hG4bK0bBac8c2c3cb90659df From: sip:7173381800@192.168.27.72;isup-oli=0;tag=gK0b66d829 To: sip:+14104159270@67.231.4.93;tag=as0850c6db Call-ID: 285991942_79966325@192.168.27.72 CSeq: 297 BYE [snip] If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP debugging enabled : written to log ??
Can you post your logger.conf ? I have: [logfiles] ;console = notice,warning,error,verbose console = warning,error,verbose messages = notice,warning,error,verbose ;console = notice,warning,error,debug full = notice,warning,error,debug,verbose,dtmf when I do SIP debug I can see the SIP debug in /var/log/full. - Original Message - From: jonas kellens To: Asterisk Mailing Sent: Sunday, October 18, 2009 10:24 Subject: [asterisk-users] SIP debugging enabled : written to log ?? Hey list ! When SIP debugging is enabled I don't want to sit down and constantly look at the CLI to debug and understand what happens. Is al this debug-informatie for SIP and/or IAX written to a log file ? I have 3 logfiles : debug, verbose and messages in logger.conf but they do not contain the SIP debugging information. Is there a way to create a logfile for SIP and/or IAX debug information ? Jonas. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP debugging enabled : written to log
I have the following in /etc/asterisk/logger.conf : debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error verbose = verbose ;full = notice,warning,error,debug,verbose When I enable SIP and/or IAX debugging on the CLI and watch the logfiles, then I conclude the following : - debugging is not written to the debug-logfile - debugging is written to the verbose-logfile What I actually want is thus the debug-information separated in one SIP and/or IAX-debug logfile : debug.iax.systemname and debug.sip.systemname or something like this... On Mon, 2009-10-19 at 08:27 +1200, Neeraj Chand wrote: There was a presentation at astricon by Clod, that covers just this CLI Filters What this does is show only the filters that you set on asterisk cli, and your /var/log/asterisk/full log file also only contains the filtered output. I believe it would have been handier to have filtering, but with everything going to the full log so that if we need to debug in greater detail / look at events outside the scope of thep filters we set, it would be available in the full logs. But thats my personal perspective. This may be useful to you. The presentation should be on astricon.net some time soon Cheers! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP debugging enabled : written to log ??
Hey list ! When SIP debugging is enabled I don't want to sit down and constantly look at the CLI to debug and understand what happens. Is al this debug-informatie for SIP and/or IAX written to a log file ? I have 3 logfiles : debug, verbose and messages in logger.conf but they do not contain the SIP debugging information. Is there a way to create a logfile for SIP and/or IAX debug information ? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP debugging enabled : written to log
There was a presentation at astricon by Clod, that covers just this CLI Filters What this does is show only the filters that you set on asterisk cli, and your /var/log/asterisk/full log file also only contains the filtered output. I believe it would have been handier to have filtering, but with everything going to the full log so that if we need to debug in greater detail / look at events outside the scope of thep filters we set, it would be available in the full logs. But thats my personal perspective. This may be useful to you. The presentation should be on astricon.net some time soon Cheers! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP Debugging to separate log file
- Original Message - From: Jason Martin [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, September 06, 2007 4:58 PM Subject: [asterisk-users] SIP Debugging to separate log file Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Try using Ngrep ngrep -t -W byline -d any -w SIP ID port 5060 Where SIP ID is the id of your sip account. It should give you everything you need. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP Debugging to separate log file
Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? Regards Bilal Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? I always find it easier to extract the SIP messaging traffic by using tcpdump or ngrep. If you use tcpdump, you can always pass the traffic through ngrep later, as well as passing it through Wireshark to get the pretty SIP traffic graphs, etc. -- Jared Smith Community Relations Manager Digium, Inc. Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP Debugging to separate log file
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? hi when iam doing this i see the server is load is very high how can i send this traffic or mirror traffic to other server and grep the reports ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP Debugging to separate log file
Dear Ram; You are able to send it for a file? Regards Bilal Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? hi when iam doing this i see the server is load is very high how can i send this traffic or mirror traffic to other server and grep the reports ram Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP Debugging to separate log file
Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP Debugging to separate log file
On Thu, 2007-09-06 at 09:58 -0400, Jason Martin wrote: Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? I always find it easier to extract the SIP messaging traffic by using tcpdump or ngrep. If you use tcpdump, you can always pass the traffic through ngrep later, as well as passing it through Wireshark to get the pretty SIP traffic graphs, etc. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP debugging
Andres wrote: Hi Klaus, The response to a CANCEL should be a 487 Request Terminated, not a 200 OK. Maybe your innovaphone Server is to blame. Hi Andres. No. The reply to the CANCEL is a 200 Ok. The reply to the cancelled INVITE is a 487. regards klaus ___ --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] SIP debugging
Kevin P. Fleming wrote: Klaus Darilion wrote: Shouldn't there be some error indication if Asterisk discards a response? Probably, although it's not clear here that Asterisk actually discarded anything. Without seeing the entire dialog, there's no way to be sure whether there were multiple Call-IDs, multiple tags, etc. involved. The problem is caused be a forked call with pedantic=yes. Asterisk --SIP-- Proxy ---SIP Sipura \ --- Cisco phone The SIPURA sends the first 180 Ringing back. Then, Asterisk ignores the responses from the Cisco phone (180+200). When setting pedantic=no, it works (I guess with pedantic=no Asterisk does not check the To tag (ugly)). Is Asterisk not able of handling multiple early dialogs with pedantic=yes? regards Klaus PS: Following the call flows pedantic=yes: -- Executing Set(Zap/50-1, [EMAIL PROTECTED]) in new stack -- Executing GotoIf(Zap/50-1, 0?103:3) in new stack -- Goto (frompbx,059966366102,3) -- Executing SetCIDNum(Zap/50-1, 00431234600265) in new stack -- Executing Dial(Zap/50-1, SIP/[EMAIL PROTECTED]|90) in new stack -- parse_srv: SRV mapped to host sip.at43.at, port 5060 We're at 213.174.230.213 port 10392 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (NAT) to 83.136.32.160:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 18 May 2006 09:31:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 293 v=0 o=root 9803 9803 IN IP4 213.174.230.213 s=session c=IN IP4 213.174.230.213 t=0 0 m=audio 10392 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called [EMAIL PROTECTED] poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 180 Ringing t: sip:[EMAIL PROTECTED];tag=f1d48eba29dc7f4i0 f: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 i: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 Record-Route: sip:[EMAIL PROTECTED]:5065,sip:83.136.32.160;ftag=as6ce265a8;lr=on Server: Sipura/SPA2000-3.1.2(NTb) Contact: sip:[EMAIL PROTECTED]:5065 Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/enum.at43.at-3323 is ringing poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:25 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on ontent-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:36 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on Content-Type: application/sdp Content-Length: 196 v=0 o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21 s=SIP Call c=IN IP4 83.136.33.21 t=0 0 m=audio 21174 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 9 lines)--- Destroying call '[EMAIL PROTECTED]' poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:36 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on Content-Type: application/sdp Content-Length: 196 v=0 o=Cisco-SIPUA 14377
Re: [Asterisk-Users] SIP debugging
Klaus Darilion wrote: Is Asterisk not able of handling multiple early dialogs with pedantic=yes? Asterisk is not capable of handling multiple dialogs in response to an outbound INVITE at all. The code is not prepared for requests that it sends to be forked by a proxy. The next major version of chan_sip (to be worked on during the next development cycle) will probably be able to handle this, but today, it's not expected to work properly. ___ --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
[Asterisk-Users] SIP debugging
Hi! I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. There is no log message, why this packet is not accepted/processed. Is there a ways to increase the sip debugging? thanks klaus Retransmitting #5 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' Retransmitting #6 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] ___ --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] SIP debugging
Klaus Darilion wrote: I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. SVN trunk is not Asterisk 1.2. There is no way to help you with this partial SIP trace, and without any Asterisk version or configuration information. Asking 'smart questions' usually leads to people being able to help you :-) ___ --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] SIP debugging
Kevin P. Fleming wrote: Klaus Darilion wrote: I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. SVN trunk is not Asterisk 1.2. Of course - sorry. I've meant Asterisk 1.2 from SVN branch 1.2 There is no way to help you with this partial SIP trace, and without any Asterisk version or configuration information. Asking 'smart questions' usually leads to people being able to help you :-) IMO this was a smart question. I did not asked to debug my call flows, but I asked how can I debug it myself. For some reason Asterisk does not like my SIP responses, but there is no Warning, Error or any other log message although verbose=9 and sip debug. Shouldn't there be some error indication if Asterisk discards a response? thanks Klaus ___ --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] SIP debugging
Klaus Darilion wrote: Shouldn't there be some error indication if Asterisk discards a response? Probably, although it's not clear here that Asterisk actually discarded anything. Without seeing the entire dialog, there's no way to be sure whether there were multiple Call-IDs, multiple tags, etc. involved. ___ --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] SIP debugging
Hi Klaus, The response to a CANCEL should be a 487 Request Terminated, not a 200 OK. Maybe your innovaphone Server is to blame. -- Andres Technical Support http://www.telesip.net Klaus Darilion wrote: Hi! I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not accept the 200 OK responses. E.g in the following example, Asterisk retransmits the CANCEL although the 200 OK is received. There is no log message, why this packet is not accepted/processed. Is there a ways to increase the sip debugging? thanks klaus Retransmitting #5 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' Retransmitting #6 (NAT) to 192.174.68.4:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- poeast01*CLI -- SIP read from 192.174.68.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport From: klaus1 sip:[EMAIL PROTECTED];tag=as4233f839 To: sip:[EMAIL PROTECTED];tag=2870350146 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: innovaphone IP800 / V6.00 dvl [06-60123] ___ --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 ___ --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
[Asterisk-Users] SIP debugging tools - Suggestions experience?
Hi, I previously posted a problem with my Zyxel P2000Wv2 wireless SIP phones and agent logins. In order to solve this problem I am looking at SIP debugging tools but I have limited experience with them. Some of the visual tools will not work as they require a software SIP phone to use and since my problem only occurs when the Zyxel phone is used and not a software SIP phone that will not work. I looked at asterisks 'sip debug' but I have not found good information about interpreting this output. Can anybody with experience in doing this make some suggestions. Also would this be something that should be posted on the developers forum??? Thanks ___ --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] Sip Debugging
Hello, When I enable SIP debugging I receive: Peer RTP is at port 10.10.60.16:0 Shouldn't the RTP port be a number between 1 - 2? - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Debugging
I have debugging on in Asterisk and sip debug. How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Debugging
It's the From: line. Mark On Thu, 6 Mar 2003, Eric Wieling wrote: I have debugging on in Asterisk and sip debug. How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
