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

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

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

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

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

Hi Kevin, Thanks for your reply. I switched to G726 32Kbps but still no luck: INVITE [SIP headers omitted] v=0 o=1 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

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

On 22 May 2009, at 16:55, Chris Maciejewski wrote: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Codec not enabled on that peer? S ___ -- Bandwidth

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

Chris Maciejewski wrote: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) 'us' does not include g726, so you have not configured your SIP user/peer to support G.726. I note Got unsupported a:fmtp

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

Yes, I was missing allow=g726 for this peer :-( Playback(/var/lib/asterisk/moh/fpm-sunshine) works OK now, however I still can't get MeetMe to work. Before I had similar problem, when MeetMe wasn't working with GSM codec because I was missing .gsm audio files. I suspect now it is the same

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

Chris Maciejewski wrote: Yes, I was missing allow=g726 for this peer :-( Playback(/var/lib/asterisk/moh/fpm-sunshine) works OK now, however I still can't get MeetMe to work. Before I had similar problem, when MeetMe wasn't working with GSM codec because I was missing .gsm audio files.

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

I do have codec_g726 loaded. As I mentioned before Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite there is only fpm-sunshine.wav file. It is only MeetMe which is not working: -- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin' (language 'en') [May 22 18:07:04]

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

Chris Maciejewski wrote: I do have codec_g726 loaded. As I mentioned before Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite there is only fpm-sunshine.wav file. It is only MeetMe which is not working: -- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin'

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

On Fri, 22 May 2009, Kevin P. Fleming wrote: This is not MeetMe, it's Playback. You specified a filename with '.slin' in it to Playback, so then Asterisk attempts to find a filename called 'entering-conf-number.slin.foo' where foo is the possible formats that Asterisk could transcode from.