[Freeswitch-users] SIP Error Message 480

Michael Collins msc at freeswitch.org
Tue Dec 15 13:27:12 PST 2009


On Tue, Dec 15, 2009 at 12:11 PM, bcxml <bcxml at hotmail.com> wrote:

>
> I have Freeswitch and Microsoft Speech Server 2007 on the same box
>
> When Speech Server initiates a call, I get a sip error message 480
>
> Here is the internal profile trace...
>
> freeswitch at HD-T2253CN>
>
> freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at
> 20:04:05
> .445011:
>   ------------------------------------------------------------------------
>   INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0
>   FROM:
> <sip:12482578002 at 127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf
> 436a29
>   TO: <sip:19059183027 at 219.175.50.104:5060;transport=tcp>
>   CSEQ: 2 INVITE
>   CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe
>   MAX-FORWARDS: 70
>   VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692
>   CONTACT:
> <sip:HD-T2253CN:1415;transport=Tcp;maddr=209.172.55.154;ms-opaque=be
> 704290e5b4e03b>;automata
>   CONTENT-LENGTH: 340
>   USER-AGENT: RTCC/3.0.0.0
>   CONTENT-TYPE: application/sdp
>   ALLOW: UPDATE
>   ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify
>
>   v=0
>   o=- 0 0 IN IP4 209.172.55.154
>   s=Microsoft Speech Server session
>   c=IN IP4 209.172.55.154
>   t=0 0
>   m=audio 35840 RTP/AVP 114 115 4 0 8 97 101
>   a=rtpmap:114 x-msrta/16000
>   a=fmtp:114 bitrate=29000
>   a=rtpmap:115 x-msrta/8000
>   a=fmtp:115 bitrate=11800
>   a=rtpmap:97 RED/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
>   ------------------------------------------------------------------------
> send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011:
>   ------------------------------------------------------------------------
>   SIP/2.0 100 Trying
>   Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431
>   FROM:
> <sip:12482578002 at 127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf
> 436a29
>   TO: <sip:19059183027 at 219.175.50.104:5060;transport=tcp>
>   CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe
>   CSEQ: 2 INVITE
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel
> sofia/inter
> nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506]
> 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing
> 12482578002-
> >19059183027 in context public
>

Are you handling "19059183027" in the public context? If so, what is that
extension doing with the call?
-MC
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