SIP Error Message 480
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@HD-T2253CN> freeswitch@HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011: ------------------------------------------------------------------------ INVITE sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@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 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /internal/12482578002@127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011:
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 To: <sip:19059183027@219.175.50.104:5060;transport=tcp>;tag=gr4aF6aS8tZ0j CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "19059183027" <sip:19059183027@209.172.55.154> ------------------------------------------------------------------------ recv 383 bytes from tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ ACK sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;tag=25bf436a29;epid=55D0 03BB53 TO: <sip:19059183027@219.175.50.104:5060;transport=tcp>;tag=gr4aF6aS8tZ0j CSEQ: 2 ACK CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTENT-LENGTH: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/ internal/12482578002@127.0.0.1:5080) Ended 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1156 Close Channel sof ia/internal/12482578002@127.0.0.1:5080 [CS_DESTROY] Can anyone point me in the right direction ? Thanks Brian -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26801000.html Sent from the Freeswitch-users mailing list archive at Nabble.com.
On Tue, Dec 15, 2009 at 12:11 PM, bcxml <bcxml@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@HD-T2253CN>
freeswitch@HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011: ------------------------------------------------------------------------ INVITE sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@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
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@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@HD-T2253CN>
freeswitch@HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011:
------------------------------------------------------------------------ INVITE sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@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@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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-- 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.
Try turning on debug logs, but from this it looks like its not matching any extensions. Mike On Dec 15, 2009, at 3:11 PM, bcxml 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...
2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel sofia/inter nal/12482578002@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 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /internal/12482578002@127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011:
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris <mike@jerris.com> wrote:
Try turning on debug logs, but from this it looks like its not matching any extensions.
Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC
Here is the link to the debug log http://pastebin.freeswitch.org/11521 Brian mercutioviz wrote:
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris <mike@jerris.com> wrote:
Try turning on debug logs, but from this it looks like its not matching any extensions.
Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC
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Yep, there is your issue.. I missed it when you pasted the extension, its a typo in your condition. Dialplan: sofia/internal/12482578002@127.0.0.1:5080 Regex (FAIL) [VoipMs] destination_number(19059183027) =~ /expression=/ break=on-false Notice what it is comparing there .. <condition field="destination_number"expression="expression="^1?(\d{10})$"> and notice the typo in your condition. Mike On Dec 15, 2009, at 9:27 PM, bcxml wrote:
Here is the link to the debug log
http://pastebin.freeswitch.org/11521
Brian
mercutioviz wrote:
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris <mike@jerris.com> wrote:
Try turning on debug logs, but from this it looks like its not matching any extensions.
Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC
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Mike..thank you so much... It works fine now Brian Michael Jerris wrote:
Yep, there is your issue.. I missed it when you pasted the extension, its a typo in your condition.
Dialplan: sofia/internal/12482578002@127.0.0.1:5080 Regex (FAIL) [VoipMs] destination_number(19059183027) =~ /expression=/ break=on-false
Notice what it is comparing there ..
<condition field="destination_number"expression="expression="^1?(\d{10})$">
and notice the typo in your condition.
Mike
On Dec 15, 2009, at 9:27 PM, bcxml wrote:
Here is the link to the debug log
http://pastebin.freeswitch.org/11521
Brian
mercutioviz wrote:
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris <mike@jerris.com> wrote:
Try turning on debug logs, but from this it looks like its not matching any extensions.
Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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-tp26801000p26805343.html Sent from the Freeswitch-users mailing list archive at Nabble.com.
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participants (3)
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bcxml
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Michael Collins
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Michael Jerris