Bart J. Smit
2006-Mar-10 07:51 UTC
[Asterisk-Users] Forward from SER to asterisk can't hang up
Hi All, I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on 5060 and asterisk on 5065. The setup is that people use serweb to create an account and register a phone. Their calls are routed from ser to asterisk and then inbound on IAX2. The server has a public and an internal interface. The real FQDN of the server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and nexphone. The pbx name is used to route calls to the asterisk. Setting up inbound calls works with this routing statement in ser.cfg: if (uri =~ "sip:[0-9]{4}@pbx\.nexusmgmt\.com") { forward(65.126.236.148,5065); break; }; The problem is that the calls are not torn down properly. Hanging up on either side does not get through to the other party. It seems like the asterisk is not accepting the BYE packet as part of the sip session. I have attached the SIP packets from an ethereal run on the external client side. The same happens if I set the forward to nmibwksip3. If I set it to pbx, the call is not set up. I have tried rewritehostport() instead of forward but this breaks the call setup too. I think that the session state breaks because the asterisk doesn't see the forwarded bye packet as part of the same session. Can I set the name(s) that asterisk answers to, same as the alias statements in ser.cfg? Will that allow me to forward to pbx instead of nmibwksip3 or IP? When I register a phone with asterisk on 5065, everything works fine. Any pointers would be very much appreciated. Thanks, Bart... -------------- next part -------------- A non-text attachment was scrubbed... Name: BYE 404 SIP packets ethereal trace Type: application/octet-stream Size: 12838 bytes Desc: BYE 404 SIP packets ethereal trace Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/4a556e89/BYE404SIPpacketsetherealtrace.obj