Hi Kevin, Thanks for your reply. I switched to G726 32Kbps but still no luck:
INVITE.... [SIP headers omitted] v=0 o=10000 1291673978 653998617 IN IP4 192.168.7.55 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8002 RTP/AVP 104 101 a=rtpmap:104 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Console SIP debug output: [May 22 16:48:20] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off Found RTP audio format 104 Found RTP audio format 101 Peer audio RTP is at port 78.105.1.131:8002 Found audio description format G726-32 for ID 104 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 22 16:48:20] NOTICE[6071]: chan_sip.c:7495 process_sdp: No compatible codecs, not accepting this offer! I note "Got unsupported a:fmtp in SDP offer" from RFC 2327: a=fmtp:<format> <format specific parameters> This attribute allows parameters that are specific to a particular format to be conveyed in a way that SDP doesn't have to understand them. The format must be one of the formats specified for the media. Format-specific parameters may be any set of parameters required to be conveyed by SDP and given unchanged to the media tool that will use this format. It is a media attribute, and is not dependent on charset. Is Twinkle sending this SDP incorrectly? Or some other issue? Thanks Chris 2009/5/22 Kevin P. Fleming <kpflem...@digium.com>: > Chris Maciejewski wrote: > >> Found unknown media description format G726-16 for ID 102 > > It's right there. > >> And asterisk is replying with "488 Not acceptable here" > > Asterisk does not support G726-16, it only supports G726-32. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users