Summary:ASTERISK-22314: Failure in canceling a call, sending OK to wrong port
Reporter:Karsten Wemheuer (kwemheuer)Labels:
Date Opened:2013-08-19 09:01:10Date Closed:2013-08-19 12:33:08
Versions: Frequency of
duplicatesASTERISK-22071 chan_sip doesn't respect Via ..completely
Environment:Debian 7.0, OpenSIPS 1.8.3Attachments:( 0) call-asterisk_1.8.22.txt
( 1) call-asterisk_1.8.23.txt
( 2) debug
Description:I've' got a problem with asterisk 1.8.23. The same scenario is working fine in 1.8.22.

Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on the same LAN, no NAT.

Phone A (Account fred) calls 456. The call is sent from the phone via the proxy (opensips) to asterisk. Asterisk calls Phone B (Account hans). The call is sent through the proxy to the second phone. The call is accepted. If one user hangs up the phone (e.g. hans), a BYE_Request is sent to asterisk. In version 1.8.22 asterisk sends a 200 OK to the phone. In version 1.8.23 asterisk send the 200 OK to the proxy. The proxy ignores the message, because of a missing second via header. The phone repeats the BYE request several times.

Attached is the debug output of asterisk and SIP-traces (ngrep) of the scenario, using asterisk version 1.8.22 (which is working) and version 1.8.23, which is not working. I wrote a mark in the traces where the issue happens ("===> This packet is sent from asterisk")

Asterisk is on, port 25060. OpenSIPS is on port 5060. Phone A (Account fred) is on, Phone B (Account hans) is on
Comments:By: Karsten Wemheuer (kwemheuer) 2013-08-19 09:02:17.243-0500

Debug output of asterisk

By: Karsten Wemheuer (kwemheuer) 2013-08-19 09:03:07.201-0500

SIP-Trace using ngrep. Working scenario with asterisk 1.8.22.

By: Karsten Wemheuer (kwemheuer) 2013-08-19 09:04:17.110-0500

SIP-Trace using ngrep. Scenario not working anymore, asterisk version 1.8.23

By: Michael L. Young (elguero) 2013-08-19 11:32:52.015-0500

Can you check out the patch (asterisk-22071-store-recvd-address.diff) on issue ASTERISK-22071 and see if that resolves this?

I have a feeling this issue might be a duplicate of that issue without digging into too much.

By: Karsten Wemheuer (kwemheuer) 2013-08-19 12:13:45.417-0500

I've checked the patch from issue [ASTERISK-22071] as Michael told. It solves my problem. The sip communications seems ok (as it was in 1.8.22).

By: Michael L. Young (elguero) 2013-08-19 12:32:39.935-0500

Thanks for your response.  We will close this issue out as a duplicate and add your comments to ASTERISK-22071.