[Freeswitch-users] How to Configure SIP DID to IVR (Maybe fixed)

Michael Jerris mike at jerris.com
Wed Jul 2 12:18:33 PDT 2008


The ERR stun failed below is killing your call.


On Jul 2, 2008, at 3:08 PM, Hristo Benev wrote:

>
> Strange I changed regex to <DID> not ^<DID> and it worked?!
>
>
>> -------- Оригинално писмо --------
>> От:  Hristo Benev
>> Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
>> До: freeswitch-users at lists.freeswitch.org
>> Изпратено на: Сряда, 2008, Юли 2 21:00:59 EEST
>
>> Here is the output:
>> ---------------------------------------
>> 2008-07-02 13:48:47 [NOTICE] switch_channel.c:533  
>> switch_channel_set_name() New Channel sofia/cisco/@ [c0d8586f- 
>> f6b9-4108-8676-c49e66f32e6d]
>> 2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()  
>> Processing ->@cisco
>> 2008-07-02 13:49:12 [ERR] sofia_glue.c:450  
>> sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org: 
>> 3478 [Timeout]
>> 2008-07-02 13:49:12 [NOTICE] mod_sofia.c:386 sofia_answer_channel()  
>> Hangup sofia/cisco/@ [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>> 2008-07-02 13:49:12 [NOTICE] switch_core_session.c:753  
>> switch_core_session_thread() Session 1 (sofia/cisco/@) Ended
>> 2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755  
>> switch_core_session_thread() Close Channel sofia/cisco/@ [CS_HANGUP]
>> ---------------------------------------
>> CallinfNumber is the number I call from
>> CiscoIP is IP of Cisco AS
>> DIDNumber is DID I have
>>
>> Thanks
>>
>> I'm doing something wrong, but what?
>> Again Here are the files
>> /conf/sip_profiles/cisco.xml (just copied external.xml and changed  
>> sip port)
>> -----------------------------------
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>> ----------------------------------------------------------
>> /conf/dialpaln/cisco.xml
>> ---------------------------------------------------------
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>> ----------------------------------------------
>> Sensitive data is obfuscated
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>>> -------- Оригинално писмо --------
>>> От:  Michael Jerris
>>> Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
>>> До: freeswitch-users at lists.freeswitch.org
>>> Изпратено на: Сряда, 2008, Юли 2 20:22:31 EEST
>>
>>> Most likely its not actually matching the extension or it runs out  
>>> of
>>> actions to perform, can you post the full debug logs from the  
>>> console?
>>>
>>> Mike
>>>
>>> On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote:
>>>
>>>>> -------- Оригинално писмо --------
>>>>> От:  Michael Jerris
>>>>> Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
>>>>> До: freeswitch-users at lists.freeswitch.org
>>>>> Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST
>>>>
>>>>> "^" seems like an invalid regex.  is that literally what
>>>>> you have there or you have some number?
>>>>>
>>>>> Mike
>>>>>
>>>>> On Jul 2, 2008, at 12:16 PM, Hristo Benev wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I'm new to FS and trying to configure DID only configuration.
>>>>>>
>>>>>> Here is the setup:
>>>>>> PSTN Cisco AS(realIP/maybe multiple ones in production)
>>>>>> FS(realIP)
>>>>>>
>>>>>> Cisco box is configured to send SIP to IP (real IP nor  
>>>>>> 192.168.x.x
>>>>>> type) and I do not have much control over it. No authentication  
>>>>>> is
>>>>>> needed.
>>>>>>
>>>>>> I'm using FS 1.0.0
>>>>>>
>>>>>> What I need to configure to send incoming PSTN calls to demo IVR
>>>>>> What I've changed?
>>>>>> Created cisco.xml file in /conf/directory/default
>>>>>> ----------------
>>>>>>
>>>>>>
>>>>>>  "/>
>>>>>>  "/>
>>>>>>  "/>
>>>>>>
>>>>>>
>>>>>> ------------------
>>>>>> Added to /conf/dialplan/default.xml
>>>>>> -----------------------------
>>>>>>
>>>>>>
>>>>>>    ">
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>> When I call DID it just rings.
>>>>>> If I connect to FS with SoftPhone on extension and I dial DID.
>>>>>>
>>>>>> I was able to get this configuration working with Asterisk(but  
>>>>>> had
>>>>>> some sound quality issues and wanted to try something else) so  
>>>>>> there
>>>>>> is no HW problem.
>>>>>>
>>>>>> Where is my misconfiguration(hopefully just this)?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> Yes there is an actual number that I do not wanted to disclose.
>>>>
>>>> I have some progress now call are accepted by FS, but something is
>>>> wrong after dialplan_hunt() is executed it hangs up.
>>>>
>>>> Thanks
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>> http://www.freeswitch.org
>>>
>>>
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>>>
>>
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>>
>
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