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Summary:ASTERISK-24548: duration of the call in error callback. Can't send 10 type frames with SIP write
Reporter:danilo borges (dbbrito)Labels:
Date Opened:2014-11-22 08:41:40.000-0600Date Closed:2014-12-30 18:48:03.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:
Versions:13.0.1 Frequency of
Occurrence		Constant
Related
Issues:
is related toASTERISK-19235 confbridge fails: chan_sip.c:6544 sip_write: Can't send 10 type frames with SIP write
Environment:Attachments:( 0) myDebugLog
( 1) sip.conf
Description:We have an error callback call duration in which he did not charge the second link (551138223465) and marks the time to 0. And it floods the cli with the error:
{noformat}
[Nov 25 09:16:01] WARNING[2595][C-00000001] chan_sip.c: Can't send 10 type frames with SIP write
{noformat}
Remembering that this link via callback is made directly to PSTN lines (not using ATA devices). The same configuration does not affect the asterisk 11 where normally use without error call duration and without error warnings. Attached hereto debug. From already thank you very much.
Comments:By: Rusty Newton (rnewton) 2014-11-24 12:50:35.579-0600

[~dbbrito] , have you seen this issue more than once?

Can you reproduce the issue at will?

If you can reproduce the issue, please provide the following:

* sip.conf or channel driver configuration for the extensions involved.
* attach an Asterisk log as instructed here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Be sure the log includes the output of "sip set debug on" and also "DEBUG" type messages in the output. You should have DEBUG verbosity turned up to 5 or above.
By: danilo borges (dbbrito) 2014-11-25 05:38:43.150-0600

Ok Rusty Newton, Annex follows the requested files. Thank you
By: Rusty Newton (rnewton) 2014-12-09 09:58:37.755-0600

Can you demonstrate via logs and dialplan where you get an incorrect call duration?

I'm not sure on the relation of the noted warnings to your call duration issues.

Frame type '10' corresponds to a Comfort Noise frame, likely sent by one of the SIP peers involved in the conference.

In general, comfort noise is not fully supported by Asterisk. You may want to turn off comfort noise generation on the peers and see if that alleviates this issue.
By: Rusty Newton (rnewton) 2014-12-30 18:48:03.073-0600

There is not enough info to identify an issue here.

I don't believe the CLI "WARNING"s are related to the issue described. We'll need more information on the call duration issue (as requested) to investigate further.

Once you get the additional information you can find a bug marshal in irc.freenode.net #asterisk-bugs or #asterisk-dev and ask them to re-open the issue. Or alternatively you may E-mail asteriskteam at digium dot com.

Thanks.