A packet capture would be most useful. Then, you could review your SDP with
your provider.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Andy Beak
Sent: Friday, July 23, 2010 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 488 Not Acceptable Here
Hi,
I'm having real difficulty in getting calls to go through with
Asterisk. I've managed to check that my SIP connection is made to my
provider. Below is an email I received from them:
----------------snip--------------------------------snip--------------------------------snip----------------
I am not certain of the reason for rejection but it has to do with the
SDP, it does not seem to be a codec issue, the response is as you have
seen:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" <sip:XXXXX at 192.168.0.14>;tag=as5c784926
To: <sip:YYYYY at 192.168.34.1>;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef661e at 192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0
There looks to be a non-standard element in your SDP that is not
supported by any of the networks.
----------------snip--------------------------------snip--------------------------------snip----------------
Which configuration file is possibly incorrect in this scenario?
What dumps are likely to be useful to me?
Thanks,
Andy
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