search for: as11a1cd82

Displaying 4 results from an estimated 4 matches for "as11a1cd82".

2020 Sep 22
2
Negotiates g729 but RTP contains g711
Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
2020 Sep 24
2
Negotiates g729 but RTP contains g711
...;rport=5060 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82 Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11 CSeq: 102 INVITE Server: Asterisk PBX 16.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:0100000000 at 52.22.22.22:5160> Content-Length: 0 <--...
2020 Sep 25
0
Negotiates g729 but RTP contains g711
...;rport=5060 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82 Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11<mailto:7030be5a09d89a9543234da051897a49 at 41.11.11.11> CSeq: 102 INVITE Server: Asterisk PBX 16.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip...
2020 Sep 25
0
Negotiates g729 but RTP contains g711
...;rport=5060 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82 Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11<mailto:7030be5a09d89a9543234da051897a49 at 41.11.11.11> CSeq: 102 INVITE Server: Asterisk PBX 16.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip...