Summary:ASTERISK-07863: Asterisk 1.2.12 multihomed one way sound
Reporter:Stamatis Kekes (skekes)Labels:
Date Opened:2006-10-03 09:33:55Date Closed:2011-06-07 14:11:58
Versions:Frequency of
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 ?


My setup is consisted of an internal network 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

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 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.