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

Hristo Benev foxb at abv.bg
Wed Jul 2 12:08:38 PDT 2008


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