# Asterisk : No compatible codecs, not accepting this offer!

**URL:** <https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820>\
**Category:** Asterisk\
**Created:** [August 23, 2016, 5:59am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820 "2016-08-23T05:59:51Z")\
**Posts on this page:** 17\
**Page:** 1

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**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 23, 2016, 5:59am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/1 "2016-08-23T05:59:51Z")

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I get this error when I am trying to use g729 codecs on Asterisk 13

When I load “module load codec\_g729.so” It load codec\_g729.so

But when I run “module show” to see all running modules, I realized the “Unknown” for codec\_g729.so while other codecs had “core” as their tag.

When I try to test Asterisk server, I got the error NOTICE[15303][C-00005d4d]: chan\_sip.c:10753 process\_sdp: No compatible codecs, not accepting this offer!

I am using Asterisk 13 [REMOVED BY ADMINISTRATOR]

When I run “uname -a” I get

4.5.7-std-2 #1 SMP Fri Jul 1 11:00:36 UTC 2016 x86\_64 x86\_64 x86\_64 GNU/Linux

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**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\
**Post date:** [August 23, 2016, 7:15am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/2 "2016-08-23T07:15:29Z")

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This error is the result of a conflict between the peer configuration and the Asterisk configuration for the peer. It is not the result of a missing codec translator. Asterisk can still pass codecs when there is no translator.

Enable SIP debugging and the DEBUG log level and look at what is being offered as against what your allow lines allow.

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 23, 2016, 7:17am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/3 "2016-08-23T07:17:55Z")

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Okay, I will do that

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 23, 2016, 11:51pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/4 "2016-08-23T23:51:25Z")

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This is what I get from the debug

— (15 headers 12 lines) —

— (15 headers 12 lines) —  
Sending to 198.101.50.4:5060 (no NAT)  
Sending to 198.101.50.4:5060 (no NAT)  
Using INVITE request as basis request - 19a05a4e7a15528b6212c45233aa3d43@198.101 .50.4  
No matching peer for ‘12057742537’ from ‘198.101.50.4:5060’  
== Using SIP RTP CoS mark 5  
Found RTP audio format 0  
Found RTP audio format 101  
Found audio description format PCMU for ID 0  
Found audio description format telephone-event for ID 101  
[Aug 24 00:06:01] NOTICE[15303][C-00005e94]: chan\_sip.c:10753 process\_sdp: No co mpatible codecs, not accepting this offer!

\<— Reliably Transmitting (no NAT) to 198.101.50.4:5060 —\>  
SIP/2.0 488 Not acceptable here  
Via: SIP/2.0/UDP 198.101.50.4:5060;branch=z9hG4bK3177e003;received=198.101.50.4; rport=5060  
From: “12057742537” sip:12057742537@198.101.50.4;tag=as724d2d96  
To: sip:15672440000@163.172.172.59;tag=as0cf3bde1  
Call-ID: [19a05a4e7a15528b6212c45233aa3d43@198.101.50.4](mailto:19a05a4e7a15528b6212c45233aa3d43@198.101.50.4)  
CSeq: 102 INVITE  
Server: Asterisk PBX 13.10.0  
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE  
Supported: replaces, timer  
Content-Length: 0

\<------------\>  
Scheduling destruction of SIP dialog ‘19a05a4e7a15528b6212c45233aa3d43@198.101.5 0.4’ in 32000 ms (Method: INVITE)

\<— SIP read from UDP:198.101.50.4:5060 —\>  
ACK [sip:15672440000@163.172.172.59](mailto:sip:15672440000@163.172.172.59) SIP/2.0  
Via: SIP/2.0/UDP 198.101.50.4:5060;branch=z9hG4bK3177e003;rport  
From: “12057742537” sip:12057742537@198.101.50.4;tag=as724d2d96  
To: sip:15672440000@163.172.172.59;tag=as0cf3bde1  
Contact: sip:12057742537@198.101.50.4  
Call-ID: [19a05a4e7a15528b6212c45233aa3d43@198.101.50.4](mailto:19a05a4e7a15528b6212c45233aa3d43@198.101.50.4)  
CSeq: 102 ACK  
User-Agent: DiDXsuPErTecSIP4  
Max-Forwards: 70  
Remote-Party-ID: “12057742537” sip:12057742537@198.101.50.4;privacy=off;screen =no  
Content-Length: 0

\<-------------\>  
— (11 headers 0 lines) —  
Really destroying SIP dialog ‘19a05a4e7a15528b6212c45233aa3d43@198.101.50.4’ Met hod: ACK

\<— SIP read from UDP:198.101.50.4:5060 —\>  
BYE [sip:15672440000@163.172.172.59](mailto:sip:15672440000@163.172.172.59) SIP/2.0  
Via: SIP/2.0/UDP 198.101.50.4:5060;branch=z9hG4bK6ad3c9c5;rport  
From: “12057742537” sip:12057742537@198.101.50.4;tag=as724d2d96  
To: sip:15672440000@163.172.172.59;tag=as0cf3bde1  
Call-ID: [19a05a4e7a15528b6212c45233aa3d43@198.101.50.4](mailto:19a05a4e7a15528b6212c45233aa3d43@198.101.50.4)  
CSeq: 103 BYE  
User-Agent: DiDXsuPErTecSIP4  
Max-Forwards: 70  
Remote-Party-ID: “12057742537” sip:12057742537@198.101.50.4;privacy=off;screen =no  
Content-Length: 0

\<-------------\>  
— (10 headers 0 lines) —  
Sending to 198.101.50.4:5060 (no NAT)

\<— Transmitting (no NAT) to 198.101.50.4:5060 —\>  
SIP/2.0 481 Call leg/transaction does not exist  
Via: SIP/2.0/UDP 198.101.50.4:5060;branch=z9hG4bK6ad3c9c5;received=198.101.50.4; rport=5060  
From: “12057742537” sip:12057742537@198.101.50.4;tag=as724d2d96  
To: sip:15672440000@163.172.172.59;tag=as0cf3bde1  
Call-ID: [19a05a4e7a15528b6212c45233aa3d43@198.101.50.4](mailto:19a05a4e7a15528b6212c45233aa3d43@198.101.50.4)  
CSeq: 103 BYE  
Server: Asterisk PBX 13.10.0  
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE  
Supported: replaces, timer  
Content-Length: 0

\<------------\>

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<div class="post-metadata">

**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\
**Post date:** [August 24, 2016, 7:14am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/5 "2016-08-24T07:14:08Z")

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That trace is incomplete.

It would appear that you have allowguest=yes and you don’t have a matching peer or user for the other side. As such you are using the settings in the general section.

They have told you they are only interested in receiving G.711 mu-law (and RFC2833 DTMF). You have obviously disabled mu-Law in your general section.

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 24, 2016, 8:57am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/6 "2016-08-24T08:57:22Z")

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Yes, you are right, in the general section, I only enabled g729 to test for g729 and in the sip trunk that takes my extension, I only accept g729 also.

When I enable other codecs such as alaw, ulaw, gsm, it works but when I force it to g729, it doesnt.  
Shouldnt g729 be the codecs that should work overall?

Also from my debug, I saw the state found peer ‘67262622’ to and IP thats when I only set the IP in the extension

[nextnumber]  
host = ipaddress

But when I do it like this

[nextnumber]  
host = dynamic

Which I think to match any IP making the call, it says no peer found for ‘777377722’ from ‘ipaddress’

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [August 24, 2016, 1:07pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/7 "2016-08-24T13:07:43Z")

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“host=dynamic” does not match any IP making the call, it tells chan\_sip that a remote device is going to REGISTER to it to tell it its IP address. If that does not happen then the entry will have no IP address to match.

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 24, 2016, 1:23pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/8 "2016-08-24T13:23:05Z")

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So for “host=dynamic” I have to register the extension in the sip.conf like [78768272]

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [August 24, 2016, 1:24pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/9 "2016-08-24T13:24:26Z")

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You don’t register the extension in sip.conf. The term “register” in SIP means for a remote device to communicate, using SIP, and register its IP address. The device has to be configured to do so. If a device isn’t meant to register then you put its IP address explicitly using the host line.

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 24, 2016, 1:33pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/10 "2016-08-24T13:33:40Z")

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Okay I understand now, I get host logic clearly now. Also in my sip.conf file, allowing alaw, ulaw, gsm and g729 makes the audio works, If I disable all codecs except g729, I get the “No compatible codecs” error.

Shouldnt g729 works across all extensions or devices?

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [August 24, 2016, 1:36pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/11 "2016-08-24T13:36:01Z")

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That controls what codecs are allowed on the Asterisk side. It’s still up to the device to support and allow the codecs.

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [August 24, 2016, 2:11pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/12 "2016-08-24T14:11:13Z")

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But I want to make sure g729 is working, that is is why I enabled it and disable others. But I have no luck in making it work, I get No compaitble codecs error, when I run module show like codec\_g729, I see it listed but it has a tag “unknown” while other codecs had “core”.

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [August 24, 2016, 2:31pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/13 "2016-08-24T14:31:53Z")

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If the device does not ask for G729 then Asterisk can not force it to use G729.

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<div class="post-metadata">

**Author:** ![lmoraes](https://avatars.discourse-cdn.com/v4/letter/l/90ced4/32.png) [@lmoraes](https://community.asterisk.org/u/lmoraes)\
**Post date:** [January 8, 2018, 8:36pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/14 "2018-01-08T20:36:09Z")

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hey oladapoadebowale,

did you solve it?

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<div class="post-metadata">

**Author:** ![oladapoadebowale](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/oladapoadebowale/32/1813_2.png) [@oladapoadebowale](https://community.asterisk.org/u/oladapoadebowale)\
**Post date:** [January 8, 2018, 11:01pm UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/15 "2018-01-08T23:01:24Z")

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Yes. I use another G729 that is compatible to my asterisk version and it worked

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [July 10, 2019, 11:40am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/16 "2019-07-10T11:40:33Z")

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Split this topic on July 10, 2019, 11:40am UTC

4 posts were split to a new topic: [No compatible codecs with opus offer](https://community.asterisk.org/t/no-compatible-codecs-with-opus-offer/80247)

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [July 10, 2019, 11:41am UTC](https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820/17 "2019-07-10T11:41:16Z")

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Closed on July 10, 2019, 11:41am UTC


