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

2009-12-15 Thread Michael Jerris
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 .. 

  

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  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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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  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
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

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Re: [Freeswitch-users] SIP Error Message 480

2009-12-15 Thread Michael Collins
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris  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 Michael Jerris
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/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:


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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



  



  

 

conf\sip_profiles\external\VoipMs.xml


  





  





mercutioviz wrote:
> 
> On Tue, Dec 15, 2009 at 12:11 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...
>>
>> 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:
>> ;epid=55D003BB53;tag=25bf
>> 436a29
>>   TO: 
>>   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:
>> > 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:
>> ;epid=55D003BB53;tag=25bf
>> 436a29
>>   TO: 
>>   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 Michael Collins
On Tue, Dec 15, 2009 at 12:11 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...
>
> 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:
> ;epid=55D003BB53;tag=25bf
> 436a29
>   TO: 
>   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:
>  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:
> ;epid=55D003BB53;tag=25bf
> 436a29
>   TO: 
>   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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