<br><br><div class="gmail_quote">On Tue, Dec 15, 2009 at 12:11 PM, bcxml <span dir="ltr"><<a href="mailto:bcxml@hotmail.com">bcxml@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
I have Freeswitch and Microsoft Speech Server 2007 on the same box<br>
<br>
When Speech Server initiates a call, I get a sip error message 480<br>
<br>
Here is the internal profile trace...<br>
<br>
freeswitch@HD-T2253CN><br>
<br>
freeswitch@HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at<br>
20:04:05<br>
.445011:<br>
------------------------------------------------------------------------<br>
INVITE sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0<br>
FROM:<br>
<sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf<br>
436a29<br>
TO: <sip:19059183027@219.175.50.104:5060;transport=tcp><br>
CSEQ: 2 INVITE<br>
CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe<br>
MAX-FORWARDS: 70<br>
VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692<br>
CONTACT:<br>
<sip:HD-T2253CN:1415;transport=Tcp;maddr=209.172.55.154;ms-opaque=be<br>
704290e5b4e03b>;automata<br>
CONTENT-LENGTH: 340<br>
USER-AGENT: RTCC/<a href="http://3.0.0.0" target="_blank">3.0.0.0</a><br>
CONTENT-TYPE: application/sdp<br>
ALLOW: UPDATE<br>
ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify<br>
<br>
v=0<br>
o=- 0 0 IN IP4 209.172.55.154<br>
s=Microsoft Speech Server session<br>
c=IN IP4 209.172.55.154<br>
t=0 0<br>
m=audio 35840 RTP/AVP 114 115 4 0 8 97 101<br>
a=rtpmap:114 x-msrta/16000<br>
a=fmtp:114 bitrate=29000<br>
a=rtpmap:115 x-msrta/8000<br>
a=fmtp:115 bitrate=11800<br>
a=rtpmap:97 RED/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431<br>
FROM:<br>
<sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf<br>
436a29<br>
TO: <sip:19059183027@219.175.50.104:5060;transport=tcp><br>
CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe<br>
CSEQ: 2 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel<br>
sofia/inter<br>
nal/<a href="http://12482578002@127.0.0.1:5080" target="_blank">12482578002@127.0.0.1:5080</a> [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506]<br>
2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing<br>
12482578002-<br>
>19059183027 in context public<br></blockquote><div><br>Are you handling "19059183027" in the public context? If so, what is that extension doing with the call?<br>-MC<br></div></div><br>