[asterisk-users] Can't get G.726 to work.

[Chris Maciejewski]

Hi,

I have both codec_g726.so and format_g726.so loaded:

root at 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