RE: [Asterisk-Users] SIP reinvite still does not occour
Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Then test, you'll see it happen. Cheers Hoa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Schreiter Sent: Friday, June 30, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP reinvite still does not occour Hi, I have in my sip.conf disallow=all allow=alaw in order to avoid any codec problems disturbing reinvite. And of course I have: canreinvite=yes In extensions.conf there is only one Dial command. It has no qualifiers like t or T. Just Dial(SIP/[EMAIL PROTECTED]) Anyway, asterisk does not try to reinvite. asterisk tells -- Attempting native bridge of SIP/01234567 ... but in the debug output there no reinvite. Using tcpdump I can see, that the audio data are going via the asterisk box in the middle, not direct between the endpoints. Is there anything else, which can prevent a reinvite? dtmp-settings? nat-settings? Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP reinvite still does not occour
Hoa Thai Duy schrieb: Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Hi, yes, to avoid transcoding problems I only have one codec, just alaw. Anything else is disallowed. That's why I don't understand, why there is no reinvite. Thanks for answering! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP reinvite still does not occour
On Fri, 2006-06-30 at 10:39 +0200, Roger Schreiter wrote: Hoa Thai Duy schrieb: Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Hi, yes, to avoid transcoding problems I only have one codec, just alaw. Anything else is disallowed. That's why I don't understand, why there is no reinvite. Thanks for answering! Iirc if you have something like a t or T in your Dial command in extensions.conf than canreinvite will not work because Asterisk has to stay in the middle to take care of the t or T. Remove these (and maybe othger) options from the Dial command and give it a try again. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
