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Summary:ASTERISK-23930: Call Barging/Whispering issue on SIP
Reporter:Govind Rajan (Govind)Labels:
Date Opened:2014-06-25 14:06:01Date Closed:2014-08-01 16:47:24
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:1.8.28.2 Frequency of
Occurrence
Related
Issues:
Environment:Tried with Ubuntu 12.04 / Asterisk 1.8.28 and CentOS-6 Asterisk 1.8.18Attachments:( 0) debug.log
Description:Hi,

Call Barging and Whispering is not happening over SIP. Instead call listening is happening when I do barge(B) or whisper(w). FYI, Everything is working when I use DAHDI(PRI lines).

I am struck on this for the last two days. I am using latest asterisk 1.8.28. When I do Barge with the option 'Bqs', Barging is not happening. I am able to listen the existing conversation. But I am not able to participate. The same issue with whisper as well.

What might be the wrong. I am attaching the sip debug the log for reference. Please help me to find out the issue.

Please find the sip log below.
[Edit by Rusty - Removed reporters inline debug - reporter attached to issue]
Comments:By: Matt Jordan (mjordan) 2014-06-25 17:16:00.178-0500

Please attach logs to this issue as files. Large log files in the issue description are a bit tough to parse and read.

By: Matt Jordan (mjordan) 2014-06-25 17:21:22.381-0500

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Govind Rajan (Govind) 2014-06-25 23:30:33.893-0500

Attaching the sip debug log file.

By: Govind Rajan (Govind) 2014-06-25 23:44:10.408-0500

FYI, I have attached the log file.

By: Rusty Newton (rnewton) 2014-06-27 07:14:35.872-0500

Nothing is apparent in the debug to me.

Can you provide the necessary, simplest configuration to reproduce the issue? That would likely be a sip.conf file, extensions.conf file and a description of the call-flow.

Does the issue still happen when you don't use either of the 's' or 'S' options?

Did this happen after upgrading? Between which versions? (Please write the complete version number)


By: Rusty Newton (rnewton) 2014-07-14 08:48:03.710-0500

[~Govind] Is the issue still occurring? Can you provide the requested information?

By: Rusty Newton (rnewton) 2014-08-01 16:47:15.953-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines