Summary:ASTERISK-12063: Zaptel channel detects hangup, but does not hangup bridged SIP channel
Reporter:Paul Belanger (pabelanger)Labels:
Date Opened:2008-05-21 11:13:49Date Closed:2008-08-13 15:44:46
Versions:Frequency of
Environment:Attachments:( 0) messages
( 1) siptrace
Description:Having an odd issue with Asterisk-1.6-beta9.

Zaptel <-> Asterisk <-> SIP

We place a call into Zaptel card, Asterisk rings SIP channel, call connects.  We hangup from Zaptel side, Asterisk detects hangup, drops zaptel, but SIP channel is still active.  Not until the call is disconnected from SIP side, does the asterisk detect the hangup.



loadzone = us


Also attached full debug message logfile
Comments:By: Jared Smith (jsmith) 2008-05-21 11:15:44

Can you give us more details, such as the CLI output (with debug messages on) and/or a packet trace?  With the limited information you've given us, it's a bit hard to tell exactly what's happening.

By: Jason Parker (jparker) 2008-05-21 11:22:14

I believe there is/was already a similar issue open for this.  It is/was caused by sip over tcp.

By: Paul Belanger (pabelanger) 2008-05-21 11:24:13

@jsmith: Both files are now uploaded.

By: Paul Belanger (pabelanger) 2008-05-21 11:26:20

@qwell: Correct, we're using SIP over TCP for this project.  I know it's still early in the game for it too.

By: Paul Belanger (pabelanger) 2008-05-31 14:39:05

Does Digium have a developer that works on SIP over TCP, or was this more a community project?

Just trying to figure out if a possible fix maybe coming soon. If not, no problems, will just have to move to plan B for a work around.


By: Raj Jain (rjain) 2008-06-01 04:53:39

Wondering if this could be due to the 'maddr' parameter presented in the Contact header in the 200 OK:

CONTACT: <sip:sv0071iv.internal.xxx:5070;transport=Tcp;maddr=>;automata

You may want to try disabling maddr if possible, or running a scenario where maddr is used in conjunction w/ UDP.

By: Paul Belanger (pabelanger) 2008-06-01 23:32:16

@rjain: Thanks for the information.  Unfortunately, our remote end SIP client does not expose the API's to modify the SIP stack.

At this point, we're only to make SIP changes on the Asterisk side of things.

If our remote end SIP client supported UDP (whole different story), I'm sure this would be a moot issue.

Thanks again,

By: Paul Belanger (pabelanger) 2008-07-11 14:00:54

Was curious if there was anything I could do to help with this issue?  I'll be more than happy to help with a patch, if somebody could point me to the location to start.

By: Brett Bryant (bbryant) 2008-07-11 14:25:52

pabelanger, I think this is a duplicate if ASTERISK-11599, i've provided a patch in that bug, would you mind testing it to see if it solves your problem?


By: Paul Belanger (pabelanger) 2008-07-11 14:35:52

bbryant: Just tested and seems to fix the issue.  Will roll into production today and update you with results.

By: Paul Belanger (pabelanger) 2008-07-11 15:01:22

bbryant: Good news, just rolled onto production machine and patch fixed the issue.  +1 for commit to trunk and 1.6 branch! :)

By: Jason Parker (jparker) 2008-08-13 15:44:44

Closing, as this is a duplicate of 17120.