Summary: | ASTERISK-07863: Asterisk 1.2.12 multihomed one way sound | ||
Reporter: | Stamatis Kekes (skekes) | Labels: | |
Date Opened: | 2006-10-03 09:33:55 | Date Closed: | 2011-06-07 14:11:58 |
Priority: | Trivial | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Today I activated my secodn DSL line and I tuned my PBX to send outgoing calls via the second DSL. I investigated the traffic and I saw that the sip control packets were travieling nice. The problem encountered in the sound. When I was calling the person that I was speaking to, it was not able to hear my voice, but they were not able to hear mine. I tried the bug ID 3143 but I was not able to patch can_sip.c Any ideas ? ****** ADDITIONAL INFORMATION ****** My setup is consisted of an internal network 10.0.0.0/24 and an external with routable IP's. | ||
Comments: | By: Serge Vecher (serge-v) 2006-10-03 10:51:44 you are describing something that looks like a configuration issue -- please seek support on the #asterisk channel on IRC or use the asterisk-users mailing list. Please read the bug guidelines before opening any new bugs. By: Stamatis Kekes (skekes) 2006-10-04 03:15:50 Well my system is multihomed and there is not a configuration issue. The same configuration in a sigle homed machine works fine. It seems that the asterisk can not route the rtp packets corectly. By: Jason Parker (jparker) 2006-10-06 13:36:23 is there a reason that this bug is private? By: Stamatis Kekes (skekes) 2006-10-09 03:57:49 Well I forgot to make it public. Is there an option to change the bug from private to public ? I would appreceate if you can change that. Best regards Stamatis By: Olle Johansson (oej) 2006-10-09 06:28:31 We need more informatino from you about this, if you consider it a bug. Read the bug guidelines and give us the required information. By: Stamatis Kekes (skekes) 2006-10-09 07:10:24 Well I run Asterisk 1.2.12.1 on a Gentoo Linux having kernel 2.6.16-gentoo-r12. I have two NIC's on for the internal and one for the extenral network. By enabling the debug as I red in this site, I found the following output: -------------------------------------------- -- Including context 'abs' in context 'default' -- Executing Goto("SIP/101-b50039d0", "abs_outgoing|8882109531283|1") in new stack -- Goto (abs_outgoing,8882109531283,1) -- Executing Dial("SIP/101-b50039d0", "SIP/2109531283@evoice2||r") in new stack -- Called 2109531283@evoice2 P[ 2] After SETUP BC -- Executing Answer("mISDN/2-2", "") in new stack P[ 2] After SETUP BC -- Executing Wait("mISDN/2-2", "1") in new stack -- SIP/evoice2-007ee390 answered SIP/101-b50039d0 -- Attempting native bridge of SIP/101-b50039d0 and SIP/evoice2-007ee390 -- Executing Playback("mISDN/2-2", "thank_you") in new stack -- Playing 'thank_you' (language 'en') -- Executing GotoIfTime("mISDN/2-2", "*|*|11-20|aug?closed|s|1") in new stack -- Executing BackGround("mISDN/2-2", "ivr-gr2") in new stack -- Playing 'ivr-gr2' (language 'en') asterisk*CLI> de debug devstate asterisk*CLI> debug 5 No such command 'debug 5' (type 'help' for help) -- Timeout on mISDN/2-2 == CDR updated on mISDN/2-2 -- Executing Dial("mISDN/2-2", " SIP/100|15|rt") in new stack Oct 9 15:07:17 WARNING[4863]: channel.c:2597 ast_request: No channel type registered for ' SIP' Oct 9 15:07:17 NOTICE[4863]: app_dial.c:1056 dial_exec_full: Unable to create channel of type ' SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Queue("mISDN/2-2", "MyQueue|t") in new stack -- Started music on hold, class 'default', on channel 'mISDN/2-2' -- Called SIP/100 -- Called SIP/101 -- Called SIP/102 -- Called SIP/104 -- Called SIP/106 -- Called SIP/107 -- Called SIP/120 -- Called SIP/121 -- SIP/102-008185d0 is ringing -- SIP/120-0082e650 is ringing -- SIP/106-00823610 is ringing -- SIP/100-0080c3b0 is ringing -- SIP/104-0081ddf0 is ringing -- SIP/107-00828e30 is ringing -- SIP/101-00815730 is ringing -- SIP/121-00833e70 is ringing -- SIP/100-0080c3b0 answered mISDN/2-2 -- Stopped music on hold on mISDN/2-2 == Spawn extension (abs_outgoing, 8882109531283, 1) exited non-zero on 'SIP/101-b50039d0' == Spawn extension (abs_incoming, t, 2) exited non-zero on 'mISDN/2-2' -------------------------------------------- I think it migh help you this. That call gave me on way audio. I was able to hear but the reciever was not able to hear me. In the internal network the SIP works perfectly but the problem exist only when I try to route the calls to my SIP provider. By: Olle Johansson (oej) 2006-10-09 08:00:23 Have you set externip and localnet? By: Stamatis Kekes (skekes) 2006-10-16 03:00:05 Yes, of course I have it but the result is the same with or without it. P.S. Sorry for being late on my answer but I was not at my office. By: Olle Johansson (oej) 2006-11-14 06:06:03.000-0600 As per the bug guidelines, we need a full SIP debug of the call. Please re-read the bug guidelines and upload a debug file. Thanks. By: Serge Vecher (serge-v) 2006-11-17 10:42:19.000-0600 1) Prepare test environment (reduce the amount of unrelated traffic on the server); 2) Make sure your logger.conf has the following line: console => notice,warning,error,debug 3) restart Asterisk with the following command: 'asterisk -Tvvvvvdddddgc | tee /tmp/verbosedebug.txt' 4) Enable SIP transaction logging with the following CLI commands: set debug 4 set verbose 4 sip debug 5) Attach verbosedebug.txt to the issue. By: Serge Vecher (serge-v) 2007-01-09 13:01:27.000-0600 Please reopen when and if you are able to produce the required debugging information from the latest 1.2 release. |