SIP/2.0 488 Not Acceptable Here

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Jun 18th, 2010
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Hello,


I'm trying to make some calls through a Cisco AS5400 to a pc with SJphone. The Cisco AS5400 config is very simple, two voip dialpeer with G711br8 codec forced.


Sjphone has a SDP error but I can't find more information of what is going on.


I launched a debug ccsip in the gateway :


*Nov 17 17:10:22.899: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP  10.30.2.66:5060;branch=z9hG4bK1411A5
From: "362635 " <sip:[email protected]>;tag=9528A348-1FC8
To: <sip:[email protected]>
Date: Sun, 17 Nov 2002 17:10:22 GMT
Call-ID: [email protected]
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 1254012559-4186313174-2153627101-3806327143
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "362635 " <sip:[email protected]>;party=calling;screen=no;privacy=off
Timestamp: 1037553022
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.30.2.66:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 233


v=0
o=CiscoSystemsSIP-GW-UserAgent 9940 3109 IN IP4 10.30.2.66
s=SIP Call
c=IN IP4 10.30.2.66
t=0 0
m=audio 19104 RTP/AVP 18 19
c=IN IP4 10.30.2.66
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:19 CN/8000
a=ptime:20


*Nov 17 17:10:22.903: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.30.2.62:5060
*Nov 17 17:10:22.903: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Nov 17 17:10:22.903: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.2.66:5060;branch=z9hG4bK1411A5;received=10.30.2.66
From: "362635 " <sip:[email protected]>;tag=9528A348-1FC8
To: "unknown" <sip:[email protected]>;tag=26243c4571
Call-ID: [email protected]
CSeq: 101 INVITE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)
Timestamp: 1037553022




*Nov 17 17:10:22.903: //46/000000000000/SIP/State/sipSPIChangeState: 0xC122D0B8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
*Nov 17 17:10:22.903: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.30.2.62:5060
*Nov 17 17:10:22.903: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Nov 17 17:10:22.907: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.30.2.66:5060;branch=z9hG4bK1411A5;received=10.30.2.66
From: "362635 " <sip:[email protected]>;tag=9528A348-1FC8
To: "unknown" <sip:[email protected]>;tag=26243c4571
Call-ID: [email protected]
CSeq: 101 INVITE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)


Thanks for your help!

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cpembleton Mon, 06/21/2010 - 09:17
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"488 Not Acceptable Here" tells the peer that the codec(s) offered are not supported.  Per the INVITE you are offering G729B.  SJPhone only supports G729 with the paid version.  Not sure if the paid version will support the g729b though, you would have to check with them. 


Try setting your codec to g711ulaw (north america/Japan) or g711alaw (Everywhere else) and see if that works.  There is a setting in the SJphone that lists the available codecs listed for use.  Check with there support guides.


HTH


Thanks,

Chad

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