[Freeswitch-users] Ringing after call has been rejected

2009-12-18 Thread bcxml 

I have an incomming call being answered by FreeSwitch and passed to IVR
application which rejects the call. 

The call is never answered by FreeSwitch, but instead of hearing a busy
signal, the caller hears ringing.

Can anyone advise how I can get the user to hear a busy signal after call
rejection instead of ringing.

Here is the debug trace 

http://pastebin.freeswitch.org/11558

Thanks


Brian

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[Freeswitch-users] SIP Error Message 480

2009-12-15 Thread bcxml 

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...

freeswi...@hd-t2253cn

freeswi...@hd-t2253cn recv 958 bytes from tcp/[209.172.55.154]:1431 at
20:04:05
.445011:
   
   INVITE sip:19059183...@219.175.50.104:5060;transport=tcp SIP/2.0
   FROM:
sip:12482578...@127.0.0.1:5080;transport=tcp;epid=55D003BB53;tag=25bf
436a29
   TO: sip:19059183...@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:12482578...@127.0.0.1:5080;transport=tcp;epid=55D003BB53;tag=25bf
436a29
   TO: sip:19059183...@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/12482578...@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/12482578...@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:12482578...@127.0.0.1:5080;transport=tcp;epid=55D003BB53;tag=25bf
436a29
   To: sip:19059183...@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:19059183...@209.172.55.154

   
recv 383 bytes from tcp/[209.172.55.154]:1431 at 20:04:05.445011:
   
   ACK sip:19059183...@219.175.50.104:5060;transport=tcp SIP/2.0
   FROM:
sip:12482578...@127.0.0.1:5080;transport=tcp;tag=25bf436a29;epid=55D0
03BB53
   TO: sip:19059183...@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/12482578...@127.0.0.1:5080) Ended
2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1156 Close Channel
sof
ia/internal/12482578...@127.0.0.1:5080 [CS_DESTROY]

Can anyone point me in the right direction ?

Thanks

Brian
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Re: [Freeswitch-users] SIP Error Message 480

2009-12-15 Thread 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 bc...@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...

 freeswi...@hd-t2253cn

 freeswi...@hd-t2253cn recv 958 bytes from tcp/[209.172.55.154]:1431 at
 20:04:05
 .445011:
  
 
   INVITE sip:19059183...@219.175.50.104:5060;transport=tcp SIP/2.0
   FROM:
 sip:12482578...@127.0.0.1:5080;transport=tcp;epid=55D003BB53;tag=25bf
 436a29
   TO: sip:19059183...@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:12482578...@127.0.0.1:5080;transport=tcp;epid=55D003BB53;tag=25bf
 436a29
   TO: sip:19059183...@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/12482578...@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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Re: [Freeswitch-users] SIP Error Message 480

2009-12-15 Thread bcxml 


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 m...@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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Re: [Freeswitch-users] SIP Error Message 480

2009-12-15 Thread bcxml 


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/12482578...@127.0.0.1:5080 Regex (FAIL) [VoipMs]
 destination_number(19059183027) =~ /expression=/ break=on-false
 
 Notice what it is comparing there .. 
 
   condition
 field=destination_numberexpression=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 m...@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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[Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread bcxml 

I am very new to Freeswitch so please accept my appologies if these questions
seem to be trivial 

I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I
have been successful in getting Freeswitch to pass an incomming PSTN call to
Speech Server. But I cannot get Freeswitch to dial out a call or transfer a
call that is sent from Speech Server 

I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called
voipms.xml which contains the following..(I have an account with voip.ms) 

include 
  gateway name=voipms





  /gateway
 /include 

And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called
Outbound.xml which contains the following 

extension name=Outbound 
  condition field=destination_number expression=^(1{0,1}\d{10})$ 
action application=set
data=effective_caller_id_number=1222333/ 
action application=bridge data=sofia/gateway/voipms/$1/ 
  /condition 
/extension 
  
When my Speech Server application tries to get FreeSwitch to transfer to
another number, the console shows the following 


2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/+1 
9059183...@199.173.95.16:5060 to xml[%23904161...@public] 
2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/interna 
l/2482578...@127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 

2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing
+14165551212 
-%23904161234 in context public

2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2
(sofia/ 
internal/2482578...@127.0.0.1:5060) Ended 
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel
sof 
ia/internal/2482578...@127.0.0.1:5060 [CS_DESTROY] 
2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia 
/external/+19059183...@199.173.95.16:5060 [CS_EXECUTE] [NORMAL_CLEARING] 
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/ 
external/+19059183...@199.173.95.16:5060) Ended 
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel
sof 
ia/external/+19059183...@199.173.95.16:5060 [CS_DESTROY] 
  
I really dont understand the line above that I have in Bold  Italic 

The number being transfered to was 4161234567...so I would have thought the
line should read.. 

Processing +14165551212-4161234567 in context public 

Can anyone tell me what the %2390 means and also any problems with my XML
files that could be preventing the transfers from taking place 

Thanks 

Brian
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Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound

2009-12-11 Thread bcxml 

The version is

FreeSWITCH Version 1.0.4 (14460)


Brian


Brian West-3 wrote:
 
 %23 is # so the question is should we URL decode that before routing?   
 I thought we did... what version are you using now?
 
 /b
 
 On Dec 11, 2009, at 5:34 PM, Michael Collins wrote:
 
 This line is basically saying that you have a call coming from  
 4165551212 and it's looking for a destination number of  
 %23904161234. The key here is that it is coming in the public  
 context so you'll need to handle the routing in conf/dialplan/ 
 public.xml

 What should this call be doing once it comes in to FS?

 -MC
 
 
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