[Freeswitch-users] SIP Error Message 480

[bcxml]

I have the following setup....

conf\dialplan\public\VoipMs.xml

<include>
    <extension name="VoipMs">
      <condition field="destination_number"
expression="expression="^1?(\d{10})$">
        <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="bridge" data="sofia/gateway/VoipMs/1$1"/>
      </condition>
    </extension>
</include> 

conf\sip_profiles\external\VoipMs.xml

<include>
  <gateway name="VoipMs">
    
    
    
    
    
  </gateway>
</include>




mercutioviz wrote:
> 
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26802352.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.