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:[EMAIL PROTECTED]") {
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...
BYE 404 SIP packets ethereal trace
Description: BYE 404 SIP packets ethereal trace
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