Can you paste a sip debug by chance, some CLI output? I'd love to see what's actually happening.
- Joshua Colp. (file in #asterisk on Freenode) -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, June 07, 2005 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 180 Ringing? (BUG?) Mirko Marghitola wrote: > Asterisk don't send the "180 Ringing" SIP message to the calling phone > when the called party is ringing. How can I force asterisk to send the > ringing messages? The option 'r' in the Dial() command or the > Ringing() command didn't solve the problem. > Mirko > Did the sip channel driver sent a progress when the called phone started ringing? In this case the driver does not send the ringing. Anyway, I don't think this behavior is correct because it breaks other protocols. E.g. if two Asterisks use SIP for their interconnection and talk H.323 with foreign gateways, then the H.323 conversation produced by the conversion H.323 <-> SIP <-> H.323 is wrong because the ALERTING of the first H.323 leg won't be generated on the second leg. And according to the H.323 recommendation ALERTING is a mandatory message. Michael. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
