Hi, I have both codec_g726.so and format_g726.so loaded:
r...@test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0 o=10000 1615261284 506628667 IN IP4 192.168.7.55 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8002 RTP/AVP 102 101 a=rtpmap:102 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Console shows: [May 22 10:29:34] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off Found RTP audio format 102 Found RTP audio format 101 Peer audio RTP is at port 78.105.1.131:8002 Found unknown media description format G726-16 for ID 102 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=0x0 (nothing)/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 10:29:34] NOTICE[6071]: chan_sip.c:7495 process_sdp: No compatible codecs, not accepting this offer! And asterisk is replying with "488 Not acceptable here" Any help and suggestions very much appreciated. Regards, Chris _______________________________________________ -- 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
