[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 Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP register
Diego Andrés Asenjo González 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/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
