# Codec can't be set to G729 **URL:** \ **Category:** Asterisk Support\ **Created:** [April 20, 2009, 1:07pm UTC](https://community.asterisk.org/t/codec-cant-be-set-to-g729/26647 "2009-04-20T13:07:30Z")\ **Posts on this page:** 4\ **Page:** 1
**Author:** ![emrom](https://avatars.discourse-cdn.com/v4/letter/e/aca169/32.png) [@emrom](https://community.asterisk.org/u/emrom)\ **Post date:** [April 20, 2009, 1:07pm UTC](https://community.asterisk.org/t/codec-cant-be-set-to-g729/26647/1 "2009-04-20T13:07:30Z")
Hi all, I’m trying to call a peer with the G729A codec, but between Asterisk and my gateway/peer the SDP header doesn’t even have “G729”. Here’s my setup: ```auto ``` Softphone1\<—\>Asterisk has no issue negotiating G729 (as initiated by the softphone), but Asterisk\<—\>Peer1 negotiate with codecs I have not allowed (GSM and G711). My sip.conf looks like this: ```auto [general] context=default fromdomain=192.168.0.14:5060 canreinvite=no [peer1] bindport=5060 port=5060 type=friend context=Inbound host=192.168.0.42 defaultip=192.168.0.14 defaultport=5060 qualify=yes disallow=all allow=G729 [softphone1] type=friend context=from-softphone host=dynamic qualify=yes fromdomain=192.168.0.14:5060 callerid=0223456789 <0223456789> disallow=all allow=G729 ``` My extensions.conf looks like this: ```auto [Inbound] ; appels entrant exten => 0223456789,1,Set(SIP_CODEC=G729) exten => 0223456789,n,Dial(SIP/softphone1) [from-softphone] exten => 2123,1,Set(SIP_CODEC=G729) exten => 2123,n,Dial(SIP/0623456789@192.168.0.42,20,rt) exten => 2123,n,Hangup() ``` I activated SIP debug on peer1, and this is what I got: [code]\*CLI\> – Executing [2123@from-softphone:1] Set(“SIP/softphone1-b6415cd0”, “SIP\_CODEC=G729”) in new stack – Executing [2123@from-softphone:2] Dial(“SIP/softphone1-b6415cd0”, “SIP/0623456789@192.168.0.42|20|rt”) in new stack Audio is at 192.168.0.14 port 13930 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 Reliably Transmitting (no NAT) to 192.168.0.42:5060: INVITE [sip:0623456789@192.168.0.42](mailto:sip:0623456789@192.168.0.42) SIP/2.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1bbee8f0;rport From: “0223456789” sip:0223456789@192.168.0.14:5060;tag=as5d5f585b To: sip:0623456789@192.168.0.42 Contact: sip:0223456789@192.168.0.14 Call-ID: 1fe2687b179c507b468ce81b2184110e@192.168.0.14:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 20 Apr 2009 10:39:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 23620 23620 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 13930 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 - - - - a=ptime:20 a=sendrecv * * * … – SIP/192.168.0.42-09a4f528 answered SIP/softphone1-b6415cd0 [Apr 20 14:52:28] NOTICE[31306]: chan\_sip.c:3659 try\_suggested\_sip\_codec: Changing codec to ‘G729’ for this call because of ${SIP\_CODEC} variable [Apr 20 14:52:28] NOTICE[31306]: chan\_sip.c:3659 try\_suggested\_sip\_codec: Changing codec to ‘G729’ for this call because of ${SIP\_CODEC} variable [/code] What bothers me in particular is that none of the codecs added to the SDP in the above debug are those which I suggested. I looked this up quite a bit before asking here, which is why I set “canreinvite=no” but because I’m using peer1 as a gateway I guess it doesn’t matter. Any ideas? Thanks! ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [April 20, 2009, 1:34pm UTC](https://community.asterisk.org/t/codec-cant-be-set-to-g729/26647/2 "2009-04-20T13:34:17Z")
You haven’t built G.729 support into your Asterisk binaries or you don’t have enough G.729 licences. ---
**Author:** ![emrom](https://avatars.discourse-cdn.com/v4/letter/e/aca169/32.png) [@emrom](https://community.asterisk.org/u/emrom)\ **Post date:** [April 20, 2009, 3:35pm UTC](https://community.asterisk.org/t/codec-cant-be-set-to-g729/26647/3 "2009-04-20T15:35:16Z")
Thanks for your quick reply David. I will look into it. ---
**Author:** ![emrom](https://avatars.discourse-cdn.com/v4/letter/e/aca169/32.png) [@emrom](https://community.asterisk.org/u/emrom)\ **Post date:** [April 21, 2009, 1:36pm UTC](https://community.asterisk.org/t/codec-cant-be-set-to-g729/26647/4 "2009-04-21T13:36:24Z")
It would appear that I would call my gateway’s IP directly as opposed to its nominal SIP profile name…The fix is: Change ```auto ``` to ```auto ``` Thanks for your reply.