[asterisk-users] SIP 406 error - cause?

[Michelle Dupuis]

I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster).  The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
 
I have attached the SIP debug output below.  It looks like codecs overlaps -
can anyone see why the call is being refused?
 
(Note that I'm not registering with the remote SIP device, just sending
directly to it by IP address).
 
Thanks,
Michelle
 
----------------------------------------------------------------------------
------------
 
Audio is at 99.99.26.93 port 16738
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 100.100.116.29:5060:
INVITE sip:1123459913134831424 at voicemaster.domain.net SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" <sip:Unknown at 99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424 at voicemaster.domain.net>
Contact: <sip:Unknown at 99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd at 99.99.26.93
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Feb 2007 21:24:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 467
 
v=0
o=root 5921 5921 IN IP4 99.99.26.93
s=session
c=IN IP4 99.99.26.93
t=0 0
m=audio 16738 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
---
    -- Called 1123459913134831424 at voicemaster.domain.net
 
<--- SIP read from 100.100.116.29:5060 --->
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" <sip:unknown at 99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424 at voicemaster.domain.net>
Contact: <sip:unknown at 99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd at 99.99.26.93
CSeq: 102 INVITE
User-agent: Asterisk PBX
Max-Forwards: 69
Date: Wed, 21 Feb 2007 21:24:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
<------------->
--- (13 headers 0 lines) ---
    -- Got SIP response 406 "Not Acceptable" back from 100.100.116.29
Transmitting (no NAT) to 100.100.116.29:5060:
ACK sip:1123459913134831424 at voicemaster.domain.net SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" <sip:Unknown at 99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424 at voicemaster.domain.net>
Contact: <sip:Unknown at 99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd at 99.99.26.93
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0