[Asterisk-Users] problems with mediatrix 1204 FXO

[Zac Sprackett]

> That worked like a champ.  Now I'm facing another issue:
> 
> When a sip phone terminates the call, asterisk sends the BYE to the 
> wrong SIP URI...  Instead of sending it to pstnline2 at 172.20.16.7
> it sends it to 611 at 172.20.16.7
> 
> Unless I'm misuderstanding the rfc asterisk is not properly 
> handling the CONTACT header.


Sorry I left out some very important details in this email.  The end
result is that asterisk is sending the bye to the wrong sip URI and
receiving a 404 response.

This causes asterisk to keep the RTP channel open even after the call
is supposed to be torn down as seen here:

carbon*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
172.20.16.7      (None)      736d2dcb343  00101/201153740  00000ms  0000ms  UNKN
172.20.16.7      (None)      4c98f793676  00101/1293907020  00000ms  0000ms  UNKN
172.20.16.7      (None)      4be294900ba  00101/1744544499  00000ms  0000ms  UNKN
3 active SIP channel(s)

The PSTN Gateway also keeps the connection active as it hasn't
received a BYE request.

Thanks again for all your help
-z