Problem forcing G729

[Nortelguy]

Ok so I bought 30 G729 licenses for my box they are installed property and seem to be working. For some reason every sip call is using ulaw as the codec no matter what I put in extensions.conf or sip.conf . At the end of the log you will see it gets rejected for not coming in G729. I had the provider do this to aid in testing. once I know the calls go thru its working.

here are the configs:

Verbosity is at least 21
dialer*CLI> show g729
0/0 encoders/decoders of 30 licensed channels are currently in use
dialer*CLI>


sip.conf:
[SIPtrunk]
disallow=all
allow=g729
allow=gsm
allow=g723
host=69.42.117.220
type=friend
canreinvite=no
dtmf=rfc2833



extensions.conf: (I added the setvar line after all else failed, still didn't work)
; dial a long distance outbound number
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls
exten => _9NXXNXXXXXX,1,setvar(SIP_CODEC=g729)
exten => _9NXXNXXXXXX,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,3,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _9NXXNXXXXXX,4,Hangup

; local area code
exten => _91305NXXXXXX,1,setvar(SIP_CODEC=g729)
exten => _91305NXXXXXX,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _91305NXXXXXX,3,Dial(${TRUNK}/${EXTEN:2},,To)
exten => _91305NXXXXXX,4,Hangup

; local area code
exten => _91786NXXXXXX,1,setvar(SIP_CODEC=g729)
exten => _91786NXXXXXX,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _91786NXXXXXX,3,Dial(${TRUNK}/${EXTEN:2},,To)
exten => _91786NXXXXXX,4,Hangup


logs from an actual call:
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing SetVar("Local/98174415815@default-900d,2", "SIP_CODEC=g729") in new stack
-- Executing AGI("Local/98174415815@default-900d,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/98174415815@default-900d,2", "SIP/69.42.117.220/8174415815||To") in new stack
We're at 192.168.3.53 port 11472
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 69.42.117.220:5060:
INVITE sip:8174415815@69.42.117.220 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.53:5060;branch=z9hG4bK22f0a5ab;rport
From: "V1210155907000153156" <sip:8882738205@192.168.3.53>;tag=as50bd6ac5
To: <sip:8174415815@69.42.117.220>
Contact: <sip:8882738205@192.168.3.53>
Call-ID: 6baf5870300f94a45998dfb43c2a65ca@192.168.3.53
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Dec 2008 20:59:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 2352 2352 IN IP4 192.168.3.53
s=session
c=IN IP4 192.168.3.53
t=0 0
m=audio 11472 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 69.42.117.220/8174415815
dialer*CLI>
<-- SIP read from 69.42.117.220:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.3.53:5060;branch=z9hG4bK54c99972;received=192.168.3.53;rport=5060
From: "V1210155907000153154" <sip:8882738205@192.168.3.53>;tag=as0a5af41b
To: <sip:8173417543@69.42.117.220>;tag=as361e1966
Call-ID: 05174fbf6fc9059e403243167f11d6c4@192.168.3.53
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


--- (10 headers 0 lines) ---

[mflorell]

Does your carrier support G729?

[Nortelguy]

of course I had it set just to that codec on their end

[mflorell]

Did you try connecting a SIP phone(hard or soft) and forcing only G729 on that same connection?

Since you bought the license from Digium I believe you are entitled to some G729 support from them if you still can't get it working.

[Nortelguy]

Did you try connecting a SIP phone(hard or soft) and forcing only G729 on that same connection?

Since you bought the license from Digium I believe you are entitled to some G729 support from them if you still can't get it working.

I tried with x-lite, .. doesn't come with G729 to test with :/ No ideas though ? Any configuration I might have missed?

[mflorell]

Have you tried Zoiper?

[Nortelguy]

Have you tried Zoiper?

No? But I'll D/L it and give it a try... I'm assuming G729 comes with it?

[mflorell]

Sorry, forgot G729 version is only available on paid version of Zoiper.

[Nortelguy]

Sorry, forgot G729 version is only available on paid version of Zoiper.

Called tech support for G729 at Digium, they didn't tell me anything more then I already knew. They couldn't offer any tech support beyond the license install itself.

[Nortelguy]

got the zoiper softphone w/g729 codec, I am forcing that codec still no dice .. Anybody have any ideas ?

[mflorell]

So Zoiper is not working with G729 for you? Is this completely outside of the Asterisk server, only from Zoiper to the asterisk server, or Zoiper directly to your carrier?

[Nortelguy]

I can't make calls thru asterisk w/zoiper w/g729 set. If I call my sip provider tell him to allow ulaw again, set zoiper to ulaw I can make calls thru asterisk. The problem is asterisk and the g729 protocol. When I do a sip debug I don't see the protocol getting 'added' just GSM ulaw and alaw.

Zoiper to asterisk I see it requesting the g729 protocol and it appears to be working. Its asterisk to the provider that I don't see it requesting g729.

[mflorell]

I don't really know what to suggest at this point, that's not a problem that I've dealt with before. The only clients that I've had that used G729 trunks have used Gafachi and we didn't have any issues with calls going through as G729.

[Nortelguy]

I don't really know what to suggest at this point, that's not a problem that I've dealt with before. The only clients that I've had that used G729 trunks have used Gafachi and we didn't have any issues with calls going through as G729.

Mike found a problem with my extensions.conf rem'd the line with the IP and added TRUNK=SIP/SIPtrunk. That way it uses the codec setup and the IP that is there... My mistake

TRUNK=SIP/SIPtrunk
;TRUNK=SIP/69.42.117.220 ; 3rd Trunk interface