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