asterisk users - Feb 2007 - SIP 406 error - cause?

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@voicemaster.domain.net SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" <sip:Unknown@99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424@voicemaster.domain.net>
Contact: <sip:Unknown@99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd@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@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@99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424@voicemaster.domain.net>
Contact: <sip:unknown@99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd@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@voicemaster.domain.net SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" <sip:Unknown@99.99.26.93>;tag=as60543531
To: <sip:1123459913134831424@voicemaster.domain.net>
Contact: <sip:Unknown@99.99.26.93>
Call-ID: 6e359fb41f074c1327fd267520a49fdd@99.99.26.93
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

On 02/22/07 06:04 Michelle Dupuis said the following:
> 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?
406s are usually returned because there're no common codecs for the call. 
check the codecs available on the voicemaster.

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh@alphaque.com                (0 0)   http://www.openmalaysiablog.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do                                        |
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a
$b."  |
| done; done                                                              |
+=========================================================================+

Hi! Everyone,

Has anybody, already experienced any issue with port forward and NAT
translation into the CheckPoint Firewall?

I did setup a DMZ for asterisk be accessible from the LAN and WAN as well.
The needed ports were properly opened but even though I am not able to
authenticate into it from both ways LAN and WAN...

It looks like to be a SIP translation issue, but I am really not sure about
that...

Does anybody have anything to say to help me?

Thanks,

Alcides

On Thu, 22 Feb 2007 13:46:07 -0200, "Alcides"
<a.cremonezi@uol.com.br> wrote:
> Hi! Everyone,
> 
> Has anybody, already experienced any issue with port forward and NAT
> translation into the CheckPoint Firewall?
> 
> I did setup a DMZ for asterisk be accessible from the LAN and WAN as well.
> The needed ports were properly opened but even though I am not able to
> authenticate into it from both ways LAN and WAN...
> 
> It looks like to be a SIP translation issue, but I am really not sure
> about
> that...
> 
> Does anybody have anything to say to help me?
> 
> Thanks,
> 
> Alcides
> 
> 
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Is this any help ?

http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
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