hi, you can simply avoid this by using local ring r option in dial command. azterisk pass local ring voice to caller and will not bridge b leg audio until b leg is answered.iin Regards,
Faisal Hanif (sent from phone) Steve Davies <[email protected]> wrote: >Hi SIP Gurus, > >I've tried to find the relevant RFCs, but am struggling. I can find >the odd opinion online, but was wondering if anyone could give a >definitive answer. > >If a SIP call is initiated (INVITE) and receives either a "180 with >SDP", or a "183 with SDP", then the remote party will start to send >audio for the inband-ringing. Asterisk then passes this audio, and it >is correctly heard by the caller. > >At present, Asterisk will also start to pass back any handset audio in >return, in theory allowing a conversation to occur on an unanswered >channel if an endpoint were designed to allow this (free phonecalls >here we come!). > >My question: > >Should: >1) Asterisk block outbound audio between the 183 Progress and the 200 >OK packets? >2) Replace any outbound audio with silence? >3) Replace outbound audio with a special NULL RTP of some sort? Does that >exist? >4) Allow any audio to be sent regardless? > >I have implemented 1) at present on our test rig, but the lack of >outbound RTP causes issues with firewall state not being set-up to >allow the inbound audio. I am not sure how hard/easy it would be to do >2) as you'd need to create silence of the correct duration to replace >each audio frame. > >Thoughts please? > >Many thanks, >Steve > >-- >_____________________________________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
