Summary: | ASTERISK-22314: Failure in canceling a call, sending OK to wrong port | ||||
Reporter: | Karsten Wemheuer (kwemheuer) | Labels: | |||
Date Opened: | 2013-08-19 09:01:10 | Date Closed: | 2013-08-19 12:33:08 | ||
Priority: | Major | Regression? | Yes | ||
Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
Versions: | 1.8.23.0 | Frequency of Occurrence | Constant | ||
Related Issues: |
| ||||
Environment: | Debian 7.0, OpenSIPS 1.8.3 | Attachments: | ( 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 192.168.10.70, port 25060. OpenSIPS is on 192.168.10.70 port 5060. Phone A (Account fred) is on 192.168.10.200, Phone B (Account hans) is on 192.168.10.201. | ||||
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. |