search for: prematuremedia

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with "prematuremedia" and "progressinband". but I can not find working settings. I don&...

...flows. I've been trying to fix this for hours, does anyone have any ideas how to get this working correctly? Asterisk version is 13.25.0 The settings I think are relevant (I'm using chan_sip): (sip.conf) ignoresdpversion=yes internal_timing=yes progressinband=never silencesuppression=no prematuremedia=no (Per peer) progressinband=yes directrtpsetup=no dtmfmode=rfc2833 directmedia=no silencesuppression=no prematuremedia=no TIA Mark. -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asteris...

...quot; which claims (youtube-video) that "their" SIP server is able to provide early video (using a Grandstream 3157v2 with "preview" enabled), but I would like to have this with asterisk... I'm currently using asterisk 12.2.x. I tried with all kinds of combinations of "prematuremedia" and "progressinband" in sip.conf and many different dialplan-extension-"scripts" but to no avail... sniffing with wireshark shows me, that the caller (doorstation) is sending H.264 video but the RTP video stream is not passed on to the callee by asterisk. (establishing a...

...your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: QueueCallerAbandon is not triggering after 1.8.3.3... (Louis Carreiro) 2. ITSP Multi IPs (Claude Hayn) 3. Re: ITSP Multi IPs (Alex Balashov) 4. 1.8 and prematuremedia problem (d tbsky) 5. Re: 1.8 and prematuremedia problem (Satish Patel) 6. Re: 1.8 and prematuremedia problem (d tbsky) 7. Re: Background music during a call (Rizwan Hisham) 8. Re: OT - Which Android handset with Wifi-only ? (Olivier) 9. Re: OUTBOUND CALLER ID (mahesh katta) 10. Re:...

...re the setings bellow sufficient to implement the above method (from experience, I've gathered mixed results and I would appreciate any input that would confirm I'm on the right or the wrong track) or shall add more magic somewhere (Answer(), Progress(), ...) ? sip.conf: progressinband=yes prematuremedia=no extensions.conf exten => _X.,1,Dial(SIP/foo) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140115/b38dafc9/attachment.html>

...xtension but there is no ringback there either and that is a problem because caller think that the phone has gone dead. So, when I call 4165555555 it works fine but if I call it through the virtual PBX it fails. I tried various combinations of "Ringing" and 'r' options and "prematuremedia=no" and "progressinband=yes" but nothing seems to help. Can someone suggest a line of enquiry? Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net