No compatible codecs!

Trying to use G.726 codec, i receive this error messages:

… No compatible codecs!

I enabled SIP DEBUG on the Asterisk console
and receive this :

By the way i am running FC4.

Anybody have expenrience to share with codec g726 or speex wideband ?

*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:

0 headers, 0 lines
Urgent handler

Sip read:
INVITE sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-;rport
Max-Forwards: 70
Contact: sip:3960@10.34.6.70:9006
To: sip:5555@canamdc.ws
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 222

v=0
o=- 4383112 4383130 IN IP4 10.34.6.70
s=eyeBeam
c=IN IP4 10.34.6.70
t=0 0
m=audio 9008 RTP/AVP 5 101
a=alt:1 1 : FF590EAE 00000014 10.34.6.70 9008
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

13 headers, 10 lines
Using latest request as basis request
Sending to 10.34.6.70 : 9006 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
To: sip:5555@canamdc.ws;tag=as532e4a6e
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:5555@10.34.6.3
Proxy-Authenticate: Digest realm=“asterisk”, nonce="4ef73b6c"
Content-Length: 0

to 10.34.6.70:9006
Scheduling destruction of call ‘4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…’ in 15000 ms
Found user '3960’
Urgent handler

Sip read:
ACK sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-;rport
To: sip:5555@canamdc.ws;tag=as532e4a6e
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 ACK
Content-Length: 0

7 headers, 0 lines
Urgent handler

Sip read:
INVITE sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-;rport
Max-Forwards: 70
Contact: sip:3960@10.34.6.70:9006
To: sip:5555@canamdc.ws
From: “Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“3960”,realm=“asterisk”,nonce=“4ef73b6c”,uri="sip:5555@canamdc.ws”,response=“f131392a9be2324d6cbc2461ff77de78”,algorithm=MD5
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 222

v=0
o=- 4383112 4383130 IN IP4 10.34.6.70
s=eyeBeam
c=IN IP4 10.34.6.70
t=0 0
m=audio 9008 RTP/AVP 5 101
a=alt:1 1 : FF590EAE 00000014 10.34.6.70 9008
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

14 headers, 10 lines
Using latest request as basis request
Sending to 10.34.6.70 : 9006 (non-NAT)
Found user '3960’
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 10.34.6.70:9008
Found description format telephone-event
Capabilities: us - 0x10 (g726), peer - audio=0x20 (adpcm)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Nov 15 13:32:49 NOTICE[4406]: chan_sip.c:2792 process_sdp: No compatible codecs!Transmitting (no NAT):
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
To: sip:5555@canamdc.ws;tag=as532e4a6e
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:5555@10.34.6.3
Content-Length: 0

to 10.34.6.70:9006
Destroying call '4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…'
Urgent handler

Sip read:
ACK sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-;rport
To: sip:5555@canamdc.ws;tag=as532e4a6e
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 ACK
Content-Length: 0

7 headers, 0 lines
Destroying call '4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…'
Urgent handler :exclamation:

[quote=“ciretoblat”]Trying to use G.726 codec, i receive this error messages:

… No compatible codecs!
[/quote]
Have you installed a g726 codec? Asterisk doesn’t come with it installed. You’ll no doubt need to buy a licence and install it yourself.

Really thought g726 was for free. As specified in OREILLY’s “ASTERISK THE FUTUR OF TELEPHONY” Chapiter 8 , p145 says license is not required.

Therefore, can you tell me where can i get it, otherwise can you tell me if there’s another codec apart g711a or g711u giving very good results on high bandwith with Asterisk ?

Well that’s probably true then. But it’s not shipped with Asterisk.

No idea, sorry. But what’s wrong with g711? It’s the industry standard.

Nothing is really wrong with it. Except that 726 uses less bandwidth and in the environment where i am, it may be very interesting.

Thanks for you help anyway.