[asterisk-bugs] [Asterisk 0014693]: codec issue, related to 14511

Asterisk Bug Tracker noreply at bugs.digium.com
Sun Mar 29 12:52:00 CDT 2009


The following issue has been set as RELATED TO issue 0014511. 
====================================================================== 
http://bugs.digium.com/view.php?id=14693 
====================================================================== 
Reported By:                pabelanger
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14693
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-03-18 10:29 CDT
Last Modified:              2009-03-29 12:52 CDT
====================================================================== 
Summary:                    codec issue, related to 14511
Description: 
http://bugs.digium.com/view.php?id=14511

I was going to post a reply to the thread, but can't seem to figure
out how to re-open it once it has been closed.

Either way, I think there maybe a problem related to the codecs issue
that was reported in the issue.

If I see the following:

sip.conf
---
[general]
disallow=all

[authentication]

[hound]
host=hound
type=peer
transport=tcp,udp
promiscredir=yes
qualify=yes
allow=ulaw
---

We get the follow errors "sip_call: No audio format found to offer"

However if we modify sip.conf to the following:
[general]
disallow=all
allow=ulaw

[authentication]

[hound]
host=hound
type=peer
transport=tcp,udp
promiscredir=yes
qualify=yes
---

The error goes away.

The one thing I did notice when we seen the error was the following in the
logs.

---
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: *** Our native formats are 0x4
(ulaw)
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: *** Joint capabilities are 0x4
(ulaw)
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: *** Our capabilities are
0x0 (nothing)
[Mar 12 12:26:51] DEBUG[19467] frame.c: Could not find preferred codec
- Going for the best codec
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: *** AST_CODEC_CHOOSE
formats are 0x4 (ulaw)
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: *** Our preferred formats
from the incoming channel are 0x4 (ulaw)
[Mar 12 12:26:51] DEBUG[19467] chan_sip.c: This channel will not be
able to handle video.
---

Here is the working version

---
[Mar 12 12:30:05] DEBUG[6314] chan_sip.c: *** Our native formats are 0x4
(ulaw)
[Mar 12 12:30:05] DEBUG[6314] chan_sip.c: *** Joint capabilities are 0x4
(ulaw)
[Mar 12 12:30:05] DEBUG[6314] chan_sip.c: *** Our capabilities are 0x4
(ulaw)
[Mar 12 12:30:05] DEBUG[6314] chan_sip.c: *** AST_CODEC_CHOOSE formats
are 0x4 (ulaw)
---



======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014511 SIP REINVITE broken in 1.6 (was working...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-29 12:52 lmadsen        Relationship added       related to 0014511  
======================================================================




More information about the asterisk-bugs mailing list