[asterisk-g729] No compatible codecs g729

[Zest]

hello
I trying to get codec working but seems there is problem

I have a
Intel i7 CPU 860 @ 2.80GHz
Asterisk V1.6.2.6 running on Gentoo 64bit - Installed direct from
asterisk as tar.gz not from emerge.

first time i had successful load the modules
codec_g723-ast16-icc-glibc-x86_64-core2-sse4.so
codec_g729-ast16-icc-glibc-x86_64-core2-sse4.so

from asterisk CLI>
module load codec_g723-ast16-icc-glibc-x86_64-core2-sse4.so
module load codec_g729-ast16-icc-glibc-x86_64-core2-sse4.so
and enabled from sip.conf
as below
[111]
type=friend
host=dynamic
username=111
secret=ff111
nat=yes
qualify=yes
context=myphones
dtmfmode=rfc2833
callerid="My EXT1" <111>
disallow=all
allow=g729

but when i try to make a call i got busy tone with below error
"No compatible codecs, not accepting this offer"
then i re-install the asterisk but when i try to load the modules
again it disconnect me direct and the asterisk stop working till i
have to remove the modules from the directory
finally the asterisk able to load the module but with different
modules "codec_g729-ast16-gcc4-glibc-x86_64-pentium4.so"
but again problem come when I make a call (busy tone) and the below
error

No compatible codecs, not accepting this offer


plz help

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[Arkadi Shishlov]

On 05/11/10 19:40, Zest wrote:
> No compatible codecs, not accepting this offer

How is "core show translation" table looks like?
If codec is loaded, then enable "sip debug on" to check the peer and your own
SIP codec proposal.

[Zest]

the line of code g729 is contain numbers
also I'm using ATA 186 sip . so i had change the Audio parameters
as below

•LBRCodec: 3
•RxCodec: 3
•TxCodec: 3
•AudioMode: 0x00150015


and here translation

viper*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723 - - - - - - - - -
- - - - - - -
gsm - - 2001 2001 3000 3000 2000 5999 6999
6999 - 3999 3000 - - 4999
ulaw - 2001 - 1 1001 1001 1 4000 5000
5000 - 2000 1001 - - 3000
alaw - 2002 1 - 1002 1002 2 4001 5001
5001 - 2001 1002 - - 3001
g726aal2 - 4000 2001 2001 - 3000 2000 5999 6999
6999 - 3999 3000 - - 4999
adpcm - 2001 2 2 1001 - 1 4000 5000
5000 - 2000 1001 - - 3000
slin - 2000 1 1 1000 1000 - 3999 4999
4999 - 1999 1000 - - 2999
lpc10 - 4000 2001 2001 3000 3000 2000 - 6999
6999 - 3999 3000 - - 4999
g729 - 4000 2001 2001 3000 3000 2000 5999 -
6999 - 3999 3000 - - 4999
speex - 4000 2001 2001 3000 3000 2000 5999 6999
- - 3999 3000 - - 4999
ilbc - - - - - - - - -
- - - - - - -
g726 - 3000 1001 1001 2000 2000 1000 4999 5999
5999 - - 2000 - - 3999
g722 - 2999 1000 1000 1999 1999 999 4998 5998
5998 - 2998 - - - 1999
siren7 - - - - - - - - -
- - - - - - -
siren14 - - - - - - - - -
- - - - - - -
slin16 - 4999 3000 3000 3999 3999 2999 6998 7998
7998 - 4998 2000 - - -


and here after enable the debug sip user and make a call , with busy
tone and error



viper*CLI>
<--- SIP read from UDP:192.168.1.23:5060 --->
INVITE sip:2...@192.168.1.170;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:5060;branch=z9hG4bKa0907f34f0a680
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>
Call-ID: 16641...@192.168.1.23
CSeq: 1 INVITE
Contact: <sip:1...@192.168.1.23:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK, UPDATE
Supported: 100rel,replaces
Content-Length: 270
Content-Type: application/sdp

v=0
o=111 26200 26200 IN IP4 192.168.1.23
s=ATA186 Call
c=IN IP4 192.168.1.23
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (13 headers 12 lines) ---
Sending to 192.168.1.23 : 5060 (no NAT)
Using INVITE request as basis request - 16641...@192.168.1.23
Found peer '111' for '111' from 192.168.1.23:5060

<--- Reliably Transmitting (NAT) to 192.168.1.23:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.23:5060;branch=z9hG4bKa0907f34f0a680;received=192.168.1.23
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>;tag=as7a893ea1
Call-ID: 16641...@192.168.1.23
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="almishal.com",
nonce="3ceb1fd3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '16641...@192.168.1.23' in 6400
ms (Method: INVITE)
viper*CLI>
<--- SIP read from UDP:192.168.1.23:5060 --->
ACK sip:2...@192.168.1.170;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:5060;branch=z9hG4bKa0907f34f0a680
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>;tag=as7a893ea1
Call-ID: 16641...@192.168.1.23
CSeq: 1 ACK
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
viper*CLI>
<--- SIP read from UDP:192.168.1.23:5060 --->
INVITE sip:2...@192.168.1.170;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:5060;branch=z9hG4bKec05ce94383820f
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>
Call-ID: 16641...@192.168.1.23
CSeq: 2 INVITE
Contact: <sip:1...@192.168.1.23:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest
username="111",realm="almishal.com",nonce="3ceb1fd3",uri="sip:
2...@192.168.1.170",response="ceed5ca489174cef3726b445245c3c63"
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK, UPDATE
Supported: 100rel,replaces
Content-Length: 270
Content-Type: application/sdp

v=0
o=111 26200 26200 IN IP4 192.168.1.23
s=ATA186 Call
c=IN IP4 192.168.1.23
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.23 : 5060 (NAT)
Using INVITE request as basis request - 16641...@192.168.1.23
Found peer '111' for '111' from 192.168.1.23:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/
video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.23:16384
Looking for 222 in myphones (domain 192.168.1.170)
list_route: hop: <sip:1...@192.168.1.23:5060;user=phone;transport=udp>
viper*CLI>
<--- Transmitting (NAT) to 192.168.1.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.23:5060;branch=z9hG4bKec05ce94383820f;received=192.168.1.23
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>
Call-ID: 16641...@192.168.1.23
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Contact: <sip:2...@192.168.1.1>
Content-Length: 0


<------------>
Audio is at 192.168.1.170 port 19906
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 168.187.187.146:2771:
INVITE sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5060;branch=z9hG4bK02b04af1;rport
Max-Forwards: 70
From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
To: <sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>
Contact: <sip:1...@192.168.1.170>
Call-ID: 4012f6d90c00072e...@192.168.1.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Tue, 11 May 2010 21:05:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1404530009 1404530009 IN IP4 192.168.1.170
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.1.170
t=0 0
m=audio 19906 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (NAT) to 168.187.187.146:2771:
INVITE sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5060;branch=z9hG4bK02b04af1;rport
Max-Forwards: 70
From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
To: <sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>
Contact: <sip:1...@192.168.1.170>
Call-ID: 4012f6d90c00072e...@192.168.1.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Tue, 11 May 2010 21:05:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1404530009 1404530009 IN IP4 192.168.1.170
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.1.170
t=0 0
m=audio 19906 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 168.187.187.146:2771:
NVITE sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5060;branch=z9hG4bK02b04af1;rport
Max-Forwards: 70
From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
To: <sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>
Contact: <sip:1...@192.168.1.170>
Call-ID: 4012f6d90c00072e...@192.168.1.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Tue, 11 May 2010 21:05:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1404530009 1404530009 IN IP4 192.168.1.170
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.1.170
t=0 0
m=audio 19906 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
viper*CLI>
<--- SIP read from UDP:168.187.187.146:2771 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.170:5060;rport=5060;branch=z9hG4bK02b04af1
To: <sip:
2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>;tag=2064b234
From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
Call-ID: 4012f6d90c00072e...@192.168.1.170
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Warning: 305 devnull "SDP: Incompatible media format: no common
codec."
Content-Length: 0
viper*CLI>

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 168.187.187.146:2771:
ACK sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5060;branch=z9hG4bK02b04af1;rport
Max-Forwards: 70
From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
To: <sip:
2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>;tag=2064b234
Contact: <sip:1...@192.168.1.170>
Call-ID: 4012f6d90c00072e...@192.168.1.170
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.6
Content-Length: 0


---

<--- Reliably Transmitting (NAT) to 192.168.1.23:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
192.168.1.23:5060;branch=z9hG4bKec05ce94383820f;received=192.168.1.23
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>;tag=as4653ca91
Call-ID: 16641...@192.168.1.23
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58


<------------>
viper*CLI>
<--- SIP read from UDP:192.168.1.23:5060 --->
ACK sip:2...@192.168.1.170;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:5060;branch=z9hG4bKec05ce94383820f
From: <sip:1...@192.168.1.170;user=phone>;tag=2430617534
To: <sip:2...@192.168.1.170;user=phone>;tag=as4653ca91
Call-ID: 16641...@192.168.1.23
CSeq: 2 ACK
User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A)
Authorization: Digest
username="111",realm="almishal.com",nonce="3ceb1fd3",uri="sip:
2...@192.168.1.170",response="ceed5ca489174cef3726b445245c3c63"
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'4012f6d90c00072e...@192.168.1.170' Method: INVITE
Really destroying SIP dialog '16641...@192.168.1.23' Method: ACK
Reliably Transmitting (NAT) to 192.168.1.23:5060:
OPTIONS sip:1...@192.168.1.23:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK5ceb0c23;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.1.1>;tag=as3e9aa7dd
To: <sip:1...@192.168.1.23:5060;user=phone;transport=udp>
Contact: <sip:aste...@192.168.1.1>
Call-ID: 6db5feff239803ab...@192.168.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Tue, 11 May 2010 21:05:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Length: 0


---
viper*CLI>
<--- SIP read from UDP:192.168.1.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK5ceb0c23;rport
From: "asterisk" <sip:aste...@192.168.1.1>;tag=as3e9aa7dd
To: <sip:
1...@192.168.1.23:5060;user=phone;transport=udp>;tag=2430617534
Call-ID: 6db5feff239803ab...@192.168.1.1
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK, UPDATE
Supported: replaces
Content-Length: 270
Content-Type: application/sdp

v=0
o=111 26864 26864 IN IP4 192.168.1.23
s=ATA186 Call
c=IN IP4 192.168.1.23
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 12 lines) ---
Really destroying SIP dialog
'6db5feff239803ab...@192.168.1.1' Method: OPTIONS
viper*CLI>

[Arkadi Shishlov]

Its the X-Lite that is rejecting g729.
AFAIK, X-Lite does not support g729.

> <--- SIP read from UDP:168.187.187.146:2771 --->
> SIP/2.0 488 Not Acceptable Here
> Via: SIP/2.0/UDP 192.168.1.170:5060;rport=5060;branch=z9hG4bK02b04af1
> To: <sip:
> 2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>;tag=2064b234
> From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
> Call-ID: 4012f6d90c00072e...@192.168.1.170
> CSeq: 102 INVITE
> User-Agent: X-Lite release 1104o stamp 56125
> Warning: 305 devnull "SDP: Incompatible media format: no common
> codec."
> Content-Length: 0

[Zest]

I'm not using x-lite
I'm using "Cisco ATA 186"

anyhow problem solve
and I'm able to using the g.729
"show sip channels" during the call active
it show me the code is g.729

I solved by
after add the below Audio parameters to my sip "Cisco ATA 186"

•LBRCodec: 3
•RxCodec: 3
•TxCodec: 3
•AudioMode: 0x00150015

i have to PowerOFF and Power-ON the "Cisco ATA 186"

On May 12, 2:58 pm, Arkadi Shishlov <arkadi.shish...@gmail.com> wrote:
> Its the X-Lite that is rejecting g729.
> AFAIK, X-Lite does not support g729.
>
> On 05/12/10 00:15, Zest wrote:
>
>
>
> > Retransmitting #2 (NAT) to 168.187.187.146:2771:
> > NVITE sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.170:5060;branch=z9hG4bK02b04af1;rport
> > Max-Forwards: 70
> > From: "Jassim Mishal" <sip:1...@192.168.1.170>;tag=as2b9b5929
> > To: <sip:2...@168.187.187.146:2771;rinstance=b3306b509a21f05b>
> > Contact: <sip:1...@192.168.1.170>

> > Call-ID: 4012f6d90c00072e5155884c02be3...@192.168.1.170

> > Call-ID: 4012f6d90c00072e5155884c02be3...@192.168.1.170