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Summary:ASTERISK-14852: Asterisk generates a BYE after 15 minutes or more consistently on trunk calls
Reporter:Michael Cramer (micc)Labels:
Date Opened:2009-09-19 23:30:04Date Closed:2009-11-23 07:51:24.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-debug1.txt
Description:All calls I have made from phone to asterisk to phone works fine for long calls. No problem. But when I make a call from phone to asterisk to sip provider to asterisk to phone, I notice asterisk generates a BYE at random time, usually after 15 to 20 minutes. I've never seen it happen before 15 minutes. I've done sip debug and sip trace, neither show any other packets except the RTP traffic working perfectly, then all the sudden asterisk sends a BYE sip packet and the call drops. This happens when calling a PSTN number as well, or another asterisk server over IAX2.

****** ADDITIONAL INFORMATION ******

Running asterisk 1.6.1.6 from source on CentOS 5.3. I have applied two patches.
https://issues.asterisk.org/view.php?id=15654
https://issues.asterisk.org/view.php?id=15538
Comments:By: Leif Madsen (lmadsen) 2009-09-21 09:42:06

As per the bug guidelines at http://www.asterisk.org/developers/bug-guidelines you need to provide the necessary information when reporting SIP issues:

SIP problem?
Include debug output! Please include output from "sip debug" if you have a SIP problem. This seems obvious, but apparently is not. Set debug to 4, verbose to 4, turn on sip history and dumphistory in sip.conf and capture all output. A packet trace from ethereal will not tell us what is happening inside your Asterisk server, so that is not a replacement.

I have a suspicion something with the RTP timers is happening that is causing Asterisk to drop the call because it isn't seeing RTP traffic when it expects to.

However, we'll need to see the SIP traces as described above to be certain.

Thanks!
By: Michael Cramer (micc) 2009-09-22 00:50:29

Ok, I uploaded the end of a sip debug trace that shows everything fine and the phone registering every 160 seconds, then out of no where BYE is sent and the call ends.
By: Michael Cramer (micc) 2009-09-22 03:06:40

After installing asterisk 1.6.2.0-rc2 I tried this again and I got a new error as the call was dropped.

[2009-09-22 00:52:41] WARNING[30531]: chan_sip.c:17299 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog 'a17a4c17973ad285'. Giving up.
   -- Hungup 'IAX2/nwd1-2137'
 == Spawn extension (macro-cramerdial, s, 4) exited non-zero on 'SIP/cramer2-b6d31f00' in macro 'cramerdial'
By: Michael Cramer (micc) 2009-09-22 03:53:06

This problem is related to issue 15270. After changing udptl=yes to udptl=no, it works fine. I'm not sure why this has anything to do with non t38 calls to pstn, but it does. Maybe this issue should be linked to 15270.
By: David Woolley (davidw) 2009-09-22 05:24:11

The only BYE in that trace is inbound, not one generated by Asterisk!
By: Olle Johansson (oej) 2009-09-22 08:10:40

1) THis debug file doesn't follow what's outlined in the guidelines - the debug output is missing.

2) The BYE is sent TO asterisk from a remote end. We hangup when we're instructed. This doesn't really look like  a bug to me - but maybe you uploaded the wrong file?

Please try again!

/O
By: Leif Madsen (lmadsen) 2009-09-22 14:48:10

Closed per the reporter in issue ASTERISK-14265 -- please continue to track the issue there. Thanks!
By: David Ruggles (thedavidfactor) 2009-11-23 07:51:24.000-0600

Removed a company name from the sip debug attachment as requested by a representative of said company.