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@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@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
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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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