[Freeswitch-users] How to Configure SIP DID to IVR
Hristo Benev
foxb at abv.bg
Wed Jul 2 11:00:59 PDT 2008
Here is the output:
---------------------------------------
2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/cisco/<CallingNumber>@<CIscoIP> [c0d8586f-f6b9-4108-8676-c49e66f32e6d]
2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing <CAllingNumber>-><DIDNumber>@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/<CallingNumber>@<CiscoIP> [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/<CallingNumber>@<CicoIP>) Ended
2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/cisco/<CallingNumber>@<CiscoIP> [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)
-----------------------------------
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="cisco">
<!-- This profile is only for cisco -->
<gateways>
<X-PRE-PROCESS cmd="include" data="cisco/*.xml"/>
</gateways>
<aliases>
<alias name="cisco"/>
</aliases>
<domains>
<domain name="$${domain}" parse="true"/>
</domains>
<settings>
<param name="debug" value="5"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="cisco"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>
----------------------------------------------------------
/conf/dialpaln/cisco.xml
---------------------------------------------------------
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="cisco">
<extension name="cisco1">
<condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/>
<condition field="destination_number" expression="^xxxxxxxxxxxx$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
<extension name="cisco2">
<condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/>
<condition field="destination_number" expression="^xxxxxxxxxxxx$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
<extension name="cisco3">
<condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/>
<condition field="destination_number" expression="^xxxxxxxxxxx$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
<extension name="cisco4">
<condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/>
<condition field="destination_number" expression="^xxxxxxxxxxx$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
</context>
</include>
----------------------------------------------
Sensitive data is obfuscated
>-------- 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
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>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
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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
>>
>> _______________________________________________
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>> Freeswitch-users at lists.freeswitch.org
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>
>
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