Diego Andrés Asenjo González
2005-Nov-25 09:23 UTC
[Asterisk-Users] Problem with SIP register
Hi! I'm registering an asterisk server in a Sysmaster with a SIP account. The registration succeeds and I can establish a call that come from the Sysmaster. After around 80 seconds the Sysmaster sends a BYE SIP message and the call hang up. This does not occur to the hard/soft SIP phones registered in the sysmaster. I debug, but the only info that I can get is the BYE message. Thanks for your suggetions soving the problem. Bye. -- Diego Andr?s Asenjo Gonz?lez Universidad del Cauca Ingeniero en Electr?nica y Telecomunicaciones -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 256 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051125/07a0e7c2/signature.pgp
Diego Andr?s Asenjo Gonz?lez wrote:>Hi! > >I'm registering an asterisk server in a Sysmaster with a SIP account. >The registration succeeds and I can establish a call that come from the >Sysmaster. > >After around 80 seconds the Sysmaster sends a BYE SIP message and the >call hang up. This does not occur to the hard/soft SIP phones registered >in the sysmaster. > >I debug, but the only info that I can get is the BYE message. > >Thanks for your suggetions soving the problem. > >Bye. > >Hi, Enable SIP debug and check which peer sends BYE at first. After call establishment, can you hear voice for 80 sec.? What about RTP in this duration? -- Baris Simsek http://www.enderunix.org/simsek/