thr3ads.net - asterisk users - [asterisk-users] media not accpetable with outgoing call on cisco [Jul 2007]

If this information is useful, please help other people find it:
Share via:

laurent schweizer

2007-Jul-17 16:26 UTC

[asterisk-users] media not accpetable with outgoing call on cisco

Hello,

I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
in my ata the the GW return a media not acceptable error.

but If i add the g729 codec the all is ok.
I see in the config of the cisco where to define codec for imcoming call but
not for outgoing

*Jul 17 15:57:02.604: Received:
INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
To: <sip:0041787518551 at 192.168.0.110>
From: 021111111 <sip:021111111 at peoplefone.ch>;tag=27B98752-469CEA8A0002F2E4-5F903B30CSeq: 10 INVITE
Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
Content-Length: 250
User-Agent: OpenSER (1.2.1-notls (i386/linux))
Contact: <sip:sems at 192.168.0.107:5070>
P-MsgFlags: 0
billingid: 106
accountid: 28928
Remote-Party-ID: <sip:0445532001 at 192.168.0.106>;party=calling;id-type=subscriber;screen=yesContent-Type: application/sdp

v=0
o=MxSIP 0 198 IN IP4 192.168.0.249
s=SIP Call
c=IN IP4 200.200.100.106
t=0 0
m=audio 39318 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
a=nortpproxy:yes

*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
*Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
and no dtmf-relay match
*Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
m-line 1

*Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
audio streams
*Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for
an incoming call - Sending 488

*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Jul 17 15:57:02.608: Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
From: 021111111 <sip:021111111 at peoplefone.ch>;tag=27B98752-469CEA8A0002F2E4-5F903B30To: <sip:0041787518551 at 192.168.0.110>;tag=C0E57710-2347
Date: Tue, 17 Jul 2007 15:57:02 GMT
Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 10 INVITE
Allow-Events: telephone-event
Content-Length: 0
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070717/bf009cf7/attachment.htm

Alex Balashov

2007-Jul-17 16:54 UTC

head link

[asterisk-users] media not accpetable with outgoing call on cisco

Laurent,

   You should be able to set it with the 'codec' subcommand on the
outgoing
dial peer as well.  'codec g711ulaw' or similar.

-- Alex

On Tue, 17 Jul 2007, laurent schweizer wrote:
> Hello,
>
> I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
> in my ata the the GW return a media not acceptable error.
>
> but If i add the g729 codec the all is ok.
> I see in the config of the cisco where to define codec for imcoming call
but
> not for outgoing
>
> *Jul 17 15:57:02.604: Received:
> INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> To: <sip:0041787518551 at 192.168.0.110>
> From: 021111111 <sip:021111111 at peoplefone.ch
>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> CSeq: 10 INVITE
> Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
> Content-Length: 250
> User-Agent: OpenSER (1.2.1-notls (i386/linux))
> Contact: <sip:sems at 192.168.0.107:5070>
> P-MsgFlags: 0
> billingid: 106
> accountid: 28928
> Remote-Party-ID: <sip:0445532001 at 192.168.0.106
>> ;party=calling;id-type=subscriber;screen=yes
> Content-Type: application/sdp
>
> v=0
> o=MxSIP 0 198 IN IP4 192.168.0.249
> s=SIP Call
> c=IN IP4 200.200.100.106
> t=0 0
> m=audio 39318 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
> SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
> and no dtmf-relay match
> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed
for
> m-line 1
>
> *Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
> audio streams
> *Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for
> an incoming call - Sending 488
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
> SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: Sent:
> SIP/2.0 488 Not Acceptable Media
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> From: 021111111 <sip:021111111 at peoplefone.ch
>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> To: <sip:0041787518551 at 192.168.0.110>;tag=C0E57710-2347
> Date: Tue, 17 Jul 2007 15:57:02 GMT
> Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 10 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>
--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671

Keshav K.

2007-Jul-18 07:12 UTC

head link

[asterisk-users] media not accpetable with outgoing call on cisco

Hi,
Your invite is going with ulaw and alaw.
need to check that what are the entries of codecs in your sip.conf, have you
allowed there ulaw and alaw or not, and next thing is if your gateway accepting,
these codecs or not.

Keshav

laurent schweizer <laurent.schweizer at gmail.com> wrote: Hello,
  
 I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in
my ata the the GW return a media not acceptable error.
  
 but If i add the g729 codec the all is ok.
 I see in the config of the cisco where to define codec for imcoming call but
not for outgoing
  
 *Jul 17 15:57:02.604: Received: 
INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5070;branch= z9hG4bK5f66.fc82e301.0
To: <sip:0041787518551 at 192.168.0.110>
From: 021111111 <sip:021111111 at peoplefone.ch
>;tag=27B98752-469CEA8A0002F2E4-5F903B30
CSeq: 10 INVITE
Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
Content-Length: 250
 User-Agent: OpenSER (1.2.1-notls (i386/linux))
Contact: <sip:sems at 192.168.0.107:5070>
P-MsgFlags: 0
billingid: 106
accountid: 28928
Remote-Party-ID: <sip:0445532001 at 192.168.0.106
>;party=calling;id-type=subscriber;screen=yes
Content-Type: application/sdp
 v=0
o=MxSIP 0 198 IN IP4 192.168.0.249
s=SIP Call
c=IN IP4 200.200.100.106
t=0 0
m=audio 39318 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000 
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
a=nortpproxy:yes
 *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
*Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec and
no dtmf-relay match
 *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
m-line 1
 *Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or audio
streams
*Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for an
incoming call - Sending 488
 *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Jul 17 15:57:02.608: Sent: 
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP  192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
From: 021111111 <sip:021111111 at
peoplefone.ch>;tag=27B98752-469CEA8A0002F2E4-5F903B30
To: < sip:0041787518551 at 192.168.0.110>;tag=C0E57710-2347
Date: Tue, 17 Jul 2007 15:57:02 GMT
Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82 
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 10 INVITE
Allow-Events: telephone-event
Content-Length: 0
 _______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

       
---------------------------------
Shape Yahoo! in your own image.  Join our Network Research Panel today!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070718/b9b3c125/attachment-0001.htm

Maybe Matching Threads

Search for more possibly parallel threads

asterisk users - Jul 2007 - media not accpetable with outgoing call on cisco

[asterisk-users] media not accpetable with outgoing call on cisco

[asterisk-users] media not accpetable with outgoing call on cisco

[asterisk-users] media not accpetable with outgoing call on cisco

Maybe Matching Threads