pfSense with Freeswitch - so far so good, but no calls going through
Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia@10.11.12.25:5060 BIND-URL sip:mod_sofia@10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1001@10.11.12.25 Contact: "1001" <sip:1001@10.11.12.145:59650;rinstance=7c2be2513804dcea;fs_nat=yes;fs_path=sip%3A1001%4010.11.12.145%3A59650%3Brinstance%3D7c2be2513804dcea> Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID: OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1002@10.11.12.25 Contact: "1002" <sip:1002@10.11.12.195:4117;rinstance=8c18309a2c957d6a;fs_nat=yes;fs_path=sip%3A1002%4010.11.12.195%3A4117%3Brinstance%3D8c18309a2c957d6a> Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 ================================================================================================= And... sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia@87.61.18.196:5080 BIND-URL sip:mod_sofia@87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 Registrations: ================================================================================================= ================================================================================================= In my Dialplan I've created these two entries: ----- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- and ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$1 I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. If I try and make a local call from 1001 to 1002 it says 2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001@10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001@1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here???
Question ---------------------------------------------- I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- Since you moved the internal profile to the lan ip address you can go ahead and dump the lan profile. Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer --- On Mon, 12/7/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Monday, December 7, 2009, 1:50 PM Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia@10.11.12.25:5060 BIND-URL sip:mod_sofia@10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1001@10.11.12.25 Contact: "1001" <sip:1001@10.11.12.145:59650;rinstance=7c2be2513804dcea;fs_nat=yes;fs_path=sip%3A1001%4010.11.12.145%3A59650%3Brinstance%3D7c2be2513804dcea> Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID: OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1002@10.11.12.25 Contact: "1002" <sip:1002@10.11.12.195:4117;rinstance=8c18309a2c957d6a;fs_nat=yes;fs_path=sip%3A1002%4010.11.12.195%3A4117%3Brinstance%3D8c18309a2c957d6a> Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 ================================================================================================= And... sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia@87.61.18.196:5080 BIND-URL sip:mod_sofia@87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 Registrations: ================================================================================================= ================================================================================================= In my Dialplan I've created these two entries: ----- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- and ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$1 I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. If I try and make a local call from 1001 to 1002 it says 2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001@10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001@1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here??? -----Inline Attachment Follows----- _______________________________________________ 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
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer
Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk>:
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer
have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist <mailinglist@fribert.dk> wrote:
Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml
Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it.
How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk>: Hi Mark
Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error.
I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch.
Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number:
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan?
Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk<http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk>> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory)
I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary?
Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier.
Question ----------------------------------------------
Extension Name musimi.dk Enabled true Order 001 Description ...
condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1
Response: ----------------------------------------------
This is correct as long as you have a gateway that is registered called musimi.dk
Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ...
action bridge sofia/internal/$
Response: ----------------------------------------------
No idea what this is for its not needed as far as I can tell.
Now please summarize what you still need help on.
Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer
_______________________________________________ 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
Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch@yahoo.com --- On Tue, 12/8/09, Nandy Dagondon <nandy1925@gmail.com> wrote: From: Nandy Dagondon <nandy1925@gmail.com> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist <mailinglist@fribert.dk> wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk>:
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console:freeswitch@firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ 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 -----Inline Attachment Follows----- _______________________________________________ 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
This is a new install, but it's grabbed from a pfSense repository.
09-12-2009 kl. 10:28 skrev Mark Crane <mctch@yahoo.com> i meddelelsen <187489.95329.qm@web56408.mail.re3.yahoo.com>:
Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch@yahoo.com --- On Tue, 12/8/09, Nandy Dagondon <nandy1925@gmail.com> wrote: From: Nandy Dagondon <nandy1925@gmail.com> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist <mailinglist@fribert.dk ( /mc/compose?to=mailinglist@fribert.dk )> wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk ( /mc/compose?to=mailinglist@fribert.dk )> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk ( /mc/compose?to=4B1DFABC020000E1000002C2@mail.fribert.dk )>:
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 ( /mc/compose?to=sofia/internal/1001@10.11.12.25 ) [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 ( /mc/compose?to=sofia/internal/1001@10.11.12.25 ) [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001@10.11.12.25 ( /mc/compose?to=sofia/internal/1001@10.11.12.25 )) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 ( /mc/compose?to=sofia/internal/1001@10.11.12.25 ) [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com ( /mc/compose?to=mctch@yahoo.com )> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com ( /mc/compose?to=659603.29094.qm@web56408.mail.re3.yahoo.com ):
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org ( /mc/compose?to=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 -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org ( /mc/compose?to=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, I have two extensions. I can even make them join a group, and if I call the group, the two extensions will ring.
08-12-2009 kl. 23:45 skrev Nandy Dagondon <nandy1925@gmail.com> i meddelelsen <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com>:
have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist <mailinglist@fribert.dk> wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk>:
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ 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
WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): <?xml version="1.0" encoding="utf-8"?> <!-- NOTICE: This context is usually accessed via authenticated callers on the sip profile on port 5060 or transfered callers from the public context which arrived via the sip profile on port 5080. Authenticated users will use the user_context variable on the user to determine what context they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml will build the domains acl using this value. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Example of doing things based on time of day. --> <extension name="tod_example" continue="true"> <!-- man strftime - M-F, 9AM to 6PM --> <condition field="${strftime(%w)}" expression="^([1-5])$"/> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="set" data="open=true"/> </condition> </extension> <extension name="global-intercept"> <condition field="destination_number" expression="^\*886$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="group-intercept"> <condition field="destination_number" expression="^\*8$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^\*870$"> <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/> </condition> </extension> <extension name="global" continue="true"> <condition field="${network_addr}" expression="^$" break="never"> <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/> <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/> </condition> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> <!-- This is an example of how to auto detect if telephone-event is missing and activate inband detection --> <!-- <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> --> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> </condition> </extension> <!-- If sip_req_host is not a local domain then this has to be an external sip uri --> <!-- <extension name="external_sip_uri" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="${outside_call}" expression="^$"/> <condition field="${domain_exists(${sip_req_host})}" expression="true"> <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/> </condition> </extension> --> <!-- snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference --> <extension name="snom-demo-2"> <condition field="destination_number" expression="^\*9001$"> <action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/> <action application="transfer" data="3000"/> </condition> </extension> <extension name="snom-demo-1"> <condition field="destination_number" expression="^\*9000$"> <!--<key> <light> <label> <user> <host> <profile> <action_name> <action>--> <action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*779$"> <action application="answer"/> <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/> <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/> <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/> <action application="eavesdrop" data="all"/> </condition> </extension> <extension name="call_return"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/> </condition> </extension> <extension name="del-group"> <condition field="destination_number" expression="^\*80(\d{2})$"> <action application="answer"/> <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/ ( 'mailto:$sip_from_user@$domain_name)"/' )> <action application="gentones" data="%(1000, 0, 320)"/> </condition> </extension> <extension name="add-group"> <condition field="destination_number" expression="^\*81(\d{2})$"> <action application="answer"/> <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/ ( 'mailto:$sip_from_user@$domain_name)"/' )> <action application="gentones" data="%(1000, 0, 640)"/> </condition> </extension> <extension name="call-group-simo"> <condition field="destination_number" expression="^\*82(\d{2})$"> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/> </condition> </extension> <extension name="call-group-order"> <condition field="destination_number" expression="^\*83(\d{2})$"> <action application="set" data="call_timeout=10"/> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="^\*8(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/ ( 'mailto:user/$dialed_extension@$domain_name"/' )> </condition> </extension> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> <!-- dial the extension 3, 4, or 5 digit extensions for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/ ( 'mailto:user/$dialed_extension@$domain_name"/' )> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="group_dial_sales"> <condition field="destination_number" expression="^\*2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/ ( 'mailto:sales@$domain_name)"/' )> </condition> </extension> <extension name="group_dial_support"> <condition field="destination_number" expression="^\*2001$"> <action application="bridge" data="group/support@${domain_name}"/ ( 'mailto:group/support@$domain_name"/' )> </condition> </extension> <extension name="group_dial_billing"> <condition field="destination_number" expression="^\*2002$"> <action application="bridge" data="group/billing@${domain_name}"/ ( 'mailto:group/billing@$domain_name"/' )> </condition> </extension> <!-- voicemail operator extension --> <!-- <extension name="operator"> <condition field="destination_number" expression="^\*operator$|^0$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="transfer" data="1000 XML features"/> </condition> </extension> --> <!-- voicemail main2 extension --> <extension name="vmain2"> <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> <extension name="send_to_voicemail_5digits"> <condition field="destination_number" expression="^\*99(\d{5})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_4digits"> <condition field="destination_number" expression="^\*99(\d{4})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_3digits"> <condition field="destination_number" expression="^\*99(\d{3})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> <!-- start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml --> <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/ ( 'mailto:$1-$domain_name@default"/' )> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^\*(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/ ( 'mailto:$1-$domain_name@wideband"/' )> </condition> </extension> <extension name="uwb_conferences"> <condition field="destination_number" expression="^\*(32\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@ultrawideband"/ ( 'mailto:$1-$domain_name@ultrawideband"/' )> </condition> </extension> <!-- MONO 48kHz conferences --> <extension name="cdquality_conferences"> <condition field="destination_number" expression="^\*(33\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@cdquality"/ ( 'mailto:$1-$domain_name@cdquality"/' )> </condition> </extension> <!-- dial the freeswitch conference via SIP--> <extension name="freeswitch_public_conf_via_sip"> <condition field="destination_number" expression="^\*9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> <!-- This will take the SAS from the b-leg and send it to the display on the a-leg phone. Known working with Polycom and Snom maybe others. --> <!-- <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/> <action application="export" data="nolocal:zrtp_secure_media=true"/> --> <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/ ( 'mailto:sofia/$use_profile/$1@conference.freeswitch.org"/' )> </condition> </extension> <!-- This extension will start a conference and invite a group. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/ ( 'mailto:madboss_intercom1@default+flagsendconf|deaf"/' )> </condition> </extension> <!-- This extension will start a conference and invite a few of people. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0912$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/> <action application="set" data="conference_auto_outcall_caller_id_number=0912"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/ ( 'mailto:madboss_intercom2@default+flagsendconf|deaf"/' )> </condition> </extension> <!--This extension will start a conference and invite several people upon entering --> <extension name="mad_boss"> <condition field="destination_number" expression="^\*0913$"> <!--These params effect the outcalls made once you join--> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>--> <!--Add as many of these as you need, These are the people you are going to call--> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss3@default"/ ( 'mailto:madboss3@default"/' )> </condition> </extension> <!-- a sample IVR --> <extension name="ivr_demo"> <condition field="destination_number" expression="^\*5000$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <!-- Create a conference on the fly and pull someone in at the same time. --> <extension name="dynamic_conference"> <condition field="destination_number" expression="^\*5001$"> <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> </condition> </extension> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^\*pagegroup$|^\*7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> <!-- Parking extensions... transferring calls to 5900 will park them in a queue. --> <extension name="park"> <condition field="destination_number" expression="^\*5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name} in"/> </condition> </extension> <!-- Parking pickup extension. Calling 5901 will pickup the call. --> <extension name="unpark"> <condition field="destination_number" expression="^\*5901$"> <action application="answer"/> <action application="fifo" data="5900@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with snom phones. Set a function key to park+lot (lot being a number or name.) Set type to Park+Orbit. You can then park and pickup using the softkey on the phone. Should work with other phones. --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="park\+(\d+)"> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- The extension is parking pickup with a to param of the fifo we are calling Some phones send things like orbit= and you can extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^parking$"/> <condition field="${sip_to_params}" expression="fifo\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with linksys phones. Set a Phone tab option Call Park Serv to yes. You can park and pickup using soft keys "park" and "unpark" found during active call when moving navigation button. The other option is to use phone's star codes (defaults to *38 and *39). --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="callpark"/> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- This extension is used with linksys phones. The extension is parking pickup with a to param of the fifo we are calling. Linksys sends orbit=<parkingslotnumber> and we extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="pickup"/> <condition field="${sip_to_params}" expression="orbit\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- Here are some examples of how to override the ringback heard by the far end. You have two variables that you can use to override this. ringback - used when a call isn't answered. (early media) transfer_ringback - used when the call is already answered. (post answer) --> <!-- Demonstration of how to override the ringback in various situations --> <extension name="wait"> <condition field="destination_number" expression="^wait$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^\*9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^\*9979$"> <action application="txfax" data="/tmp/txfax.tif"/> <action application="hangup"/> </condition> </extension> <!-- Send a 180 and let the far end generate ringback. --> <extension name="ringback_180"> <condition field="destination_number" expression="^\*9980$"> <action application="ring_ready"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <!-- Send a 183 and send uk-ring as the ringtone. (early media) --> <extension name="ringback_183_uk_ring"> <condition field="destination_number" expression="^\*9981$"> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Send a 183 and use music as the ringtone. (early media) --> <extension name="ringback_183_music_ring"> <condition field="destination_number" expression="^\*9982$"> <action application="set" data="ringback=$${hold_music}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_uk_ring"> <condition field="destination_number" expression="^\*9983$"> <action application="set" data="transfer_ringback=$${uk-ring}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_music"> <condition field="destination_number" expression="^\*9984$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <extension name="ClueCon"> <condition field="destination_number" expression="^\*9991$"> <action application="set" data="effective_caller_id_name=ClueCon"/> <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/ ( 'mailto:sofia/$$domain/brian@bkw.org"/' )> </condition> </extension> <extension name="show_info"> <condition field="destination_number" expression="^\*9992$"> <action application="answer"/> <action application="info"/> <action application="sleep" data="250"/> <action application="hangup"/> </condition> </extension> <extension name="video_record"> <condition field="destination_number" expression="^\*9993$"> <action application="answer"/> <action application="record_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="video_playback"> <condition field="destination_number" expression="^\*9994$"> <action application="answer"/> <action application="play_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="delay_echo"> <condition field="destination_number" expression="^\*9995$"> <action application="answer"/> <action application="delay_echo" data="5000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^\*9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="milliwatt"> <condition field="destination_number" expression="^\*9997$"> <action application="answer"/> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/> </condition> </extension> <extension name="tone_stream"> <condition field="destination_number" expression="^\*9998$"> <action application="answer"/> <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> </condition> </extension> <!-- You will no longer hear the bong tone. The wav file is playing stating the call is secure. The file will not play unless you have both TLS and SRTP active. --> <extension name="hold_music"> <condition field="destination_number" expression="^\*9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <!-- This really should be an IVR for zrtp enrollment but this is just a demo--> <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> <!-- You can place files in the default directory to get included. --> <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>--> <!-- WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING Anything you put below this line will usually get ignored due to the file in default/99999_enum.xml as it will transfer the call to the enum dialplan. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <extension name="enum"> <condition field="${module_exists(mod_enum)}" expression="true"/> <condition field="destination_number" expression="^(.*)$"> <action application="transfer" data="$1 enum"/> </condition> </extension> <!-- <extension name="refer"> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression> </condition> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:(.*)@(.*)>]]></expression> <action application="set" data="refer_user=$1"/> <action application="set" data="refer_domain=$2"/> <action application="info"/> <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/ ( 'mailto:sofia/$use_profile/$refer_user@$refer_domain"/' )> </condition> </extension> --> <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <!-- <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^\*7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/ ( 'mailto:ruri_contact=1001@$domain_name"/' )> <action application="execute_extension" data="ruri"/> </condition> </extension> --> <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! --> </context> </include> Then I have under default dir: musimidk.xml <extension name="musimi.dk"> <condition field="^0(.\d+)$" expression=""> <action application="bridge" data="sofia/gateway/musimi.dk/$1"/> </condition> </extension> and 9000_recordings.xml <extension name="Recordings"> <condition field="destination_number" expression="^\*(732673)$"> <action application="javascript" data="recordings.js"/> </condition> </extension> Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up???
Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): <?xml version="1.0" encoding="utf-8"?> <!-- NOTICE: This context is usually accessed via authenticated callers on the sip profile on port 5060 or transfered callers from the public context which arrived via the sip profile on port 5080. Authenticated users will use the user_context variable on the user to determine what context they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml will build the domains acl using this value. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Example of doing things based on time of day. --> <extension name="tod_example" continue="true"> <!-- man strftime - M-F, 9AM to 6PM --> <condition field="${strftime(%w)}" expression="^([1-5])$"/> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="set" data="open=true"/> </condition> </extension> <extension name="global-intercept"> <condition field="destination_number" expression="^\*886$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="group-intercept"> <condition field="destination_number" expression="^\*8$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^\*870$"> <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/> </condition> </extension> <extension name="global" continue="true"> <condition field="${network_addr}" expression="^$" break="never"> <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/> <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/> </condition> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> <!-- This is an example of how to auto detect if telephone-event is missing and activate inband detection --> <!-- <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> --> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> </condition> </extension> <!-- If sip_req_host is not a local domain then this has to be an external sip uri --> <!-- <extension name="external_sip_uri" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="${outside_call}" expression="^$"/> <condition field="${domain_exists(${sip_req_host})}" expression="true"> <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/> </condition> </extension> --> <!-- snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference --> <extension name="snom-demo-2"> <condition field="destination_number" expression="^\*9001$"> <action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/> <action application="transfer" data="3000"/> </condition> </extension> <extension name="snom-demo-1"> <condition field="destination_number" expression="^\*9000$"> <!--<key> <light> <label> <user> <host> <profile> <action_name> <action>--> <action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*779$"> <action application="answer"/> <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/> <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/> <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/> <action application="eavesdrop" data="all"/> </condition> </extension> <extension name="call_return"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/> </condition> </extension> <extension name="del-group"> <condition field="destination_number" expression="^\*80(\d{2})$"> <action application="answer"/> <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 320)"/> </condition> </extension> <extension name="add-group"> <condition field="destination_number" expression="^\*81(\d{2})$"> <action application="answer"/> <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 640)"/> </condition> </extension> <extension name="call-group-simo"> <condition field="destination_number" expression="^\*82(\d{2})$"> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/> </condition> </extension> <extension name="call-group-order"> <condition field="destination_number" expression="^\*83(\d{2})$"> <action application="set" data="call_timeout=10"/> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="^\*8(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> <!-- dial the extension 3, 4, or 5 digit extensions for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="group_dial_sales"> <condition field="destination_number" expression="^\*2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/> </condition> </extension> <extension name="group_dial_support"> <condition field="destination_number" expression="^\*2001$"> <action application="bridge" data="group/support@${domain_name}"/> </condition> </extension> <extension name="group_dial_billing"> <condition field="destination_number" expression="^\*2002$"> <action application="bridge" data="group/billing@${domain_name}"/> </condition> </extension> <!-- voicemail operator extension --> <!-- <extension name="operator"> <condition field="destination_number" expression="^\*operator$|^0$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="transfer" data="1000 XML features"/> </condition> </extension> --> <!-- voicemail main2 extension --> <extension name="vmain2"> <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> <extension name="send_to_voicemail_5digits"> <condition field="destination_number" expression="^\*99(\d{5})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_4digits"> <condition field="destination_number" expression="^\*99(\d{4})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_3digits"> <condition field="destination_number" expression="^\*99(\d{3})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> <!-- start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml --> <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^\*(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/> </condition> </extension> <extension name="uwb_conferences"> <condition field="destination_number" expression="^\*(32\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@ultrawideband"/> </condition> </extension> <!-- MONO 48kHz conferences --> <extension name="cdquality_conferences"> <condition field="destination_number" expression="^\*(33\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@cdquality"/> </condition> </extension> <!-- dial the freeswitch conference via SIP--> <extension name="freeswitch_public_conf_via_sip"> <condition field="destination_number" expression="^\*9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> <!-- This will take the SAS from the b-leg and send it to the display on the a-leg phone. Known working with Polycom and Snom maybe others. --> <!-- <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/> <action application="export" data="nolocal:zrtp_secure_media=true"/> --> <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/> </condition> </extension> <!-- This extension will start a conference and invite a group. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/> </condition> </extension> <!-- This extension will start a conference and invite a few of people. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0912$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/> <action application="set" data="conference_auto_outcall_caller_id_number=0912"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/> </condition> </extension> <!--This extension will start a conference and invite several people upon entering --> <extension name="mad_boss"> <condition field="destination_number" expression="^\*0913$"> <!--These params effect the outcalls made once you join--> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>--> <!--Add as many of these as you need, These are the people you are going to call--> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss3@default"/> </condition> </extension> <!-- a sample IVR --> <extension name="ivr_demo"> <condition field="destination_number" expression="^\*5000$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <!-- Create a conference on the fly and pull someone in at the same time. --> <extension name="dynamic_conference"> <condition field="destination_number" expression="^\*5001$"> <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> </condition> </extension> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^\*pagegroup$|^\*7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> <!-- Parking extensions... transferring calls to 5900 will park them in a queue. --> <extension name="park"> <condition field="destination_number" expression="^\*5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name} in"/> </condition> </extension> <!-- Parking pickup extension. Calling 5901 will pickup the call. --> <extension name="unpark"> <condition field="destination_number" expression="^\*5901$"> <action application="answer"/> <action application="fifo" data="5900@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with snom phones. Set a function key to park+lot (lot being a number or name.) Set type to Park+Orbit. You can then park and pickup using the softkey on the phone. Should work with other phones. --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="park\+(\d+)"> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- The extension is parking pickup with a to param of the fifo we are calling Some phones send things like orbit= and you can extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^parking$"/> <condition field="${sip_to_params}" expression="fifo\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with linksys phones. Set a Phone tab option Call Park Serv to yes. You can park and pickup using soft keys "park" and "unpark" found during active call when moving navigation button. The other option is to use phone's star codes (defaults to *38 and *39). --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="callpark"/> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- This extension is used with linksys phones. The extension is parking pickup with a to param of the fifo we are calling. Linksys sends orbit=<parkingslotnumber> and we extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="pickup"/> <condition field="${sip_to_params}" expression="orbit\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- Here are some examples of how to override the ringback heard by the far end. You have two variables that you can use to override this. ringback - used when a call isn't answered. (early media) transfer_ringback - used when the call is already answered. (post answer) --> <!-- Demonstration of how to override the ringback in various situations --> <extension name="wait"> <condition field="destination_number" expression="^wait$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^\*9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^\*9979$"> <action application="txfax" data="/tmp/txfax.tif"/> <action application="hangup"/> </condition> </extension> <!-- Send a 180 and let the far end generate ringback. --> <extension name="ringback_180"> <condition field="destination_number" expression="^\*9980$"> <action application="ring_ready"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <!-- Send a 183 and send uk-ring as the ringtone. (early media) --> <extension name="ringback_183_uk_ring"> <condition field="destination_number" expression="^\*9981$"> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Send a 183 and use music as the ringtone. (early media) --> <extension name="ringback_183_music_ring"> <condition field="destination_number" expression="^\*9982$"> <action application="set" data="ringback=$${hold_music}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_uk_ring"> <condition field="destination_number" expression="^\*9983$"> <action application="set" data="transfer_ringback=$${uk-ring}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_music"> <condition field="destination_number" expression="^\*9984$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <extension name="ClueCon"> <condition field="destination_number" expression="^\*9991$"> <action application="set" data="effective_caller_id_name=ClueCon"/> <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/> </condition> </extension> <extension name="show_info"> <condition field="destination_number" expression="^\*9992$"> <action application="answer"/> <action application="info"/> <action application="sleep" data="250"/> <action application="hangup"/> </condition> </extension> <extension name="video_record"> <condition field="destination_number" expression="^\*9993$"> <action application="answer"/> <action application="record_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="video_playback"> <condition field="destination_number" expression="^\*9994$"> <action application="answer"/> <action application="play_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="delay_echo"> <condition field="destination_number" expression="^\*9995$"> <action application="answer"/> <action application="delay_echo" data="5000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^\*9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="milliwatt"> <condition field="destination_number" expression="^\*9997$"> <action application="answer"/> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/> </condition> </extension> <extension name="tone_stream"> <condition field="destination_number" expression="^\*9998$"> <action application="answer"/> <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> </condition> </extension> <!-- You will no longer hear the bong tone. The wav file is playing stating the call is secure. The file will not play unless you have both TLS and SRTP active. --> <extension name="hold_music"> <condition field="destination_number" expression="^\*9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <!-- This really should be an IVR for zrtp enrollment but this is just a demo--> <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> <!-- You can place files in the default directory to get included. --> <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>--> <!-- WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING Anything you put below this line will usually get ignored due to the file in default/99999_enum.xml as it will transfer the call to the enum dialplan. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <extension name="enum"> <condition field="${module_exists(mod_enum)}" expression="true"/> <condition field="destination_number" expression="^(.*)$"> <action application="transfer" data="$1 enum"/> </condition> </extension> <!-- <extension name="refer"> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression> </condition> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:(.*)@(.*)>]]></expression> <action application="set" data="refer_user=$1"/> <action application="set" data="refer_domain=$2"/> <action application="info"/> <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/> </condition> </extension> --> <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <!-- <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^\*7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/> <action application="execute_extension" data="ruri"/> </condition> </extension> --> <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! --> </context> </include> Then I have under default dir: musimidk.xml <extension name="musimi.dk"> <condition field="^0(.\d+)$" expression=""> <action application="bridge" data="sofia/gateway/musimi.dk/$1"/> </condition> </extension> and 9000_recordings.xml <extension name="Recordings"> <condition field="destination_number" expression="^\*(732673)$"> <action application="javascript" data="recordings.js"/> </condition> </extension> Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ 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
Hi Mark The extensions are located under directory/default, and they look like this: <include> <user id="1001" mailbox="1001"> <params> <param name="password" value="1001"/> <param name="vm-password" value="1001"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="kenneth@fribert.dk"/ ( 'mailto:kenneth@fribert.dk"/' )> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Fribert"/> <variable name="effective_caller_id_number" value="12345678"/> </variables> </user> </include> <include> <user id="1002" mailbox="1002"> <params> <param name="password" value="1002"/> <param name="vm-password" value="1002"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="lian@fribert.dk"/ ( 'mailto:lian@fribert.dk"/' )> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1002"/> <variable name="user_context" value="default"/> </variables> </user> </include> As I understand them, the context set there, is the right one?
10-12-2009 kl. 00:00 skrev Mark Crane <mctch@yahoo.com> i meddelelsen <110796.60596.qm@web56401.mail.re3.yahoo.com>:
Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): <?xml version="1.0" encoding="utf-8"?> <!-- NOTICE: This context is usually accessed via authenticated callers on the sip profile on port 5060 or transfered callers from the public context which arrived via the sip profile on port 5080. Authenticated users will use the user_context variable on the user to determine what context they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml will build the domains acl using this value. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Example of doing things based on time of day. --> <extension name="tod_example" continue="true"> <!-- man strftime - M-F, 9AM to 6PM --> <condition field="${strftime(%w)}" expression="^([1-5])$"/> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="set" data="open=true"/> </condition> </extension> <extension name="global-intercept"> <condition field="destination_number" expression="^\*886$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="group-intercept"> <condition field="destination_number" expression="^\*8$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^\*870$"> <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/> </condition> </extension> <extension name="global" continue="true"> <condition field="${network_addr}" expression="^$" break="never"> <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/> <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/> </condition> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> <!-- This is an example of how to auto detect if telephone-event is missing and activate inband detection --> <!-- <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> --> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> </condition> </extension> <!-- If sip_req_host is not a local domain then this has to be an external sip uri --> <!-- <extension name="external_sip_uri" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="${outside_call}" expression="^$"/> <condition field="${domain_exists(${sip_req_host})}" expression="true"> <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/> </condition> </extension> --> <!-- snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference --> <extension name="snom-demo-2"> <condition field="destination_number" expression="^\*9001$"> <action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/> <action application="transfer" data="3000"/> </condition> </extension> <extension name="snom-demo-1"> <condition field="destination_number" expression="^\*9000$"> <!--<key> <light> <label> <user> <host> <profile> <action_name> <action>--> <action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*779$"> <action application="answer"/> <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/> <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/> <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/> <action application="eavesdrop" data="all"/> </condition> </extension> <extension name="call_return"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/> </condition> </extension> <extension name="del-group"> <condition field="destination_number" expression="^\*80(\d{2})$"> <action application="answer"/> <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/ ( /mc/compose?to=$%7Bsip_from_user%7D@$%7Bdomain_name%7D%29%7D%22/ )> <action application="gentones" data="%(1000, 0, 320)"/> </condition> </extension> <extension name="add-group"> <condition field="destination_number" expression="^\*81(\d{2})$"> <action application="answer"/> <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/ ( /mc/compose?to=$%7Bsip_from_user%7D@$%7Bdomain_name%7D%29%7D%22/ )> <action application="gentones" data="%(1000, 0, 640)"/> </condition> </extension> <extension name="call-group-simo"> <condition field="destination_number" expression="^\*82(\d{2})$"> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/> </condition> </extension> <extension name="call-group-order"> <condition field="destination_number" expression="^\*83(\d{2})$"> <action application="set" data="call_timeout=10"/> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="^\*8(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/ ( /mc/compose?to=user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/ )> </condition> </extension> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> <!-- dial the extension 3, 4, or 5 digit extensions for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name ( /mc/compose?to=$%7Bdialed_extension%7D@$%7Bdomain_name )} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name ( /mc/compose?to=$%7Bdialed_extension%7D@$%7Bdomain_name )} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/ ( /mc/compose?to=user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/ )> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="group_dial_sales"> <condition field="destination_number" expression="^\*2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/ ( /mc/compose?to=sales@$%7Bdomain_name%7D%29%7D%22/ )> </condition> </extension> <extension name="group_dial_support"> <condition field="destination_number" expression="^\*2001$"> <action application="bridge" data="group/support@${domain_name}"/ ( /mc/compose?to=group/support@$%7Bdomain_name%7D%22/ )> </condition> </extension> <extension name="group_dial_billing"> <condition field="destination_number" expression="^\*2002$"> <action application="bridge" data="group/billing@${domain_name}"/ ( /mc/compose?to=group/billing@$%7Bdomain_name%7D%22/ )> </condition> </extension> <!-- voicemail operator extension --> <!-- <extension name="operator"> <condition field="destination_number" expression="^\*operator$|^0$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="transfer" data="1000 XML features"/> </condition> </extension> --> <!-- voicemail main2 extension --> <extension name="vmain2"> <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> <extension name="send_to_voicemail_5digits"> <condition field="destination_number" expression="^\*99(\d{5})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_4digits"> <condition field="destination_number" expression="^\*99(\d{4})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_3digits"> <condition field="destination_number" expression="^\*99(\d{3})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> <!-- start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml --> <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/ ( /mc/compose?to=$1-$%7Bdomain_name%7D@default%22/ )> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^\*(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/ ( /mc/compose?to=$1-$%7Bdomain_name%7D@wideband%22/ )> </condition> </extension> <extension name="uwb_conferences"> <condition field="destination_number" expression="^\*(32\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@ultrawideband"/ ( /mc/compose?to=$1-$%7Bdomain_name%7D@ultrawideband%22/ )> </condition> </extension> <!-- MONO 48kHz conferences --> <extension name="cdquality_conferences"> <condition field="destination_number" expression="^\*(33\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@cdquality"/ ( /mc/compose?to=$1-$%7Bdomain_name%7D@cdquality%22/ )> </condition> </extension> <!-- dial the freeswitch conference via SIP--> <extension name="freeswitch_public_conf_via_sip"> <condition field="destination_number" expression="^\*9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> <!-- This will take the SAS from the b-leg and send it to the display on the a-leg phone. Known working with Polycom and Snom maybe others. --> <!-- <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/> <action application="export" data="nolocal:zrtp_secure_media=true"/> --> <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/ ( /mc/compose?to=sofia/$%7Buse_profile%7D/$1@conference.freeswitch.org%22/ )> </condition> </extension> <!-- This extension will start a conference and invite a group. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/ ( /mc/compose?to=madboss_intercom1@default+flags%7Bendconf%7Cdeaf%7D%22/ )> </condition> </extension> <!-- This extension will start a conference and invite a few of people. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0912$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/> <action application="set" data="conference_auto_outcall_caller_id_number=0912"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/ ( /mc/compose?to=madboss_intercom2@default+flags%7Bendconf%7Cdeaf%7D%22/ )> </condition> </extension> <!--This extension will start a conference and invite several people upon entering --> <extension name="mad_boss"> <condition field="destination_number" expression="^\*0913$"> <!--These params effect the outcalls made once you join--> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>--> <!--Add as many of these as you need, These are the people you are going to call--> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss3@default"/ ( /mc/compose?to=madboss3@default%22/ )> </condition> </extension> <!-- a sample IVR --> <extension name="ivr_demo"> <condition field="destination_number" expression="^\*5000$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <!-- Create a conference on the fly and pull someone in at the same time. --> <extension name="dynamic_conference"> <condition field="destination_number" expression="^\*5001$"> <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> </condition> </extension> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^\*pagegroup$|^\*7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> <!-- Parking extensions... transferring calls to 5900 will park them in a queue. --> <extension name="park"> <condition field="destination_number" expression="^\*5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name ( /mc/compose?to=5900@$%7Bdomain_name )} in"/> </condition> </extension> <!-- Parking pickup extension. Calling 5901 will pickup the call. --> <extension name="unpark"> <condition field="destination_number" expression="^\*5901$"> <action application="answer"/> <action application="fifo" data="5900@${domain_name ( /mc/compose?to=5900@$%7Bdomain_name )} out nowait"/> </condition> </extension> <!-- This extension is used with snom phones. Set a function key to park+lot (lot being a number or name.) Set type to Park+Orbit. You can then park and pickup using the softkey on the phone. Should work with other phones. --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="park\+(\d+)"> <action application="fifo" data="$1@${domain_name ( /mc/compose?to=$1@$%7Bdomain_name )} in undef $${hold_music}"/> </condition> </extension> <!-- The extension is parking pickup with a to param of the fifo we are calling Some phones send things like orbit= and you can extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^parking$"/> <condition field="${sip_to_params}" expression="fifo\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name ( /mc/compose?to=$1@$%7Bdomain_name )} out nowait"/> </condition> </extension> <!-- This extension is used with linksys phones. Set a Phone tab option Call Park Serv to yes. You can park and pickup using soft keys "park" and "unpark" found during active call when moving navigation button. The other option is to use phone's star codes (defaults to *38 and *39). --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="callpark"/> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression> <action application="fifo" data="$1@${domain_name ( /mc/compose?to=$1@$%7Bdomain_name )} in undef $${hold_music}"/> </condition> </extension> <!-- This extension is used with linksys phones. The extension is parking pickup with a to param of the fifo we are calling. Linksys sends orbit=<parkingslotnumber> and we extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="pickup"/> <condition field="${sip_to_params}" expression="orbit\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name ( /mc/compose?to=$1@$%7Bdomain_name )} out nowait"/> </condition> </extension> <!-- Here are some examples of how to override the ringback heard by the far end. You have two variables that you can use to override this. ringback - used when a call isn't answered. (early media) transfer_ringback - used when the call is already answered. (post answer) --> <!-- Demonstration of how to override the ringback in various situations --> <extension name="wait"> <condition field="destination_number" expression="^wait$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^\*9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^\*9979$"> <action application="txfax" data="/tmp/txfax.tif"/> <action application="hangup"/> </condition> </extension> <!-- Send a 180 and let the far end generate ringback. --> <extension name="ringback_180"> <condition field="destination_number" expression="^\*9980$"> <action application="ring_ready"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <!-- Send a 183 and send uk-ring as the ringtone. (early media) --> <extension name="ringback_183_uk_ring"> <condition field="destination_number" expression="^\*9981$"> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Send a 183 and use music as the ringtone. (early media) --> <extension name="ringback_183_music_ring"> <condition field="destination_number" expression="^\*9982$"> <action application="set" data="ringback=$${hold_music}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_uk_ring"> <condition field="destination_number" expression="^\*9983$"> <action application="set" data="transfer_ringback=$${uk-ring}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_music"> <condition field="destination_number" expression="^\*9984$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <extension name="ClueCon"> <condition field="destination_number" expression="^\*9991$"> <action application="set" data="effective_caller_id_name=ClueCon"/> <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/ ( /mc/compose?to=sofia/$$%7Bdomain%7D/brian@bkw.org%22/ )> </condition> </extension> <extension name="show_info"> <condition field="destination_number" expression="^\*9992$"> <action application="answer"/> <action application="info"/> <action application="sleep" data="250"/> <action application="hangup"/> </condition> </extension> <extension name="video_record"> <condition field="destination_number" expression="^\*9993$"> <action application="answer"/> <action application="record_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="video_playback"> <condition field="destination_number" expression="^\*9994$"> <action application="answer"/> <action application="play_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="delay_echo"> <condition field="destination_number" expression="^\*9995$"> <action application="answer"/> <action application="delay_echo" data="5000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^\*9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="milliwatt"> <condition field="destination_number" expression="^\*9997$"> <action application="answer"/> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/> </condition> </extension> <extension name="tone_stream"> <condition field="destination_number" expression="^\*9998$"> <action application="answer"/> <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> </condition> </extension> <!-- You will no longer hear the bong tone. The wav file is playing stating the call is secure. The file will not play unless you have both TLS and SRTP active. --> <extension name="hold_music"> <condition field="destination_number" expression="^\*9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <!-- This really should be an IVR for zrtp enrollment but this is just a demo--> <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> <!-- You can place files in the default directory to get included. --> <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>--> <!-- WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING Anything you put below this line will usually get ignored due to the file in default/99999_enum.xml as it will transfer the call to the enum dialplan. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <extension name="enum"> <condition field="${module_exists(mod_enum)}" expression="true"/> <condition field="destination_number" expression="^(.*)$"> <action application="transfer" data="$1 enum"/> </condition> </extension> <!-- <extension name="refer"> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression> </condition> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:(.*)@(.*)>]]></expression> <action application="set" data="refer_user=$1"/> <action application="set" data="refer_domain=$2"/> <action application="info"/> <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/ ( /mc/compose?to=sofia/$%7Buse_profile%7D/$%7Brefer_user%7D@$%7Brefer_domain%7D%22/ )> </condition> </extension> --> <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <!-- <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^\*7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/ ( /mc/compose?to=ruri_contact=1001@$%7Bdomain_name%7D%22/ )> <action application="execute_extension" data="ruri"/> </condition> </extension> --> <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! --> </context> </include> Then I have under default dir: musimidk.xml <extension name="musimi.dk"> <condition field="^0(.\d+)$" expression=""> <action application="bridge" data="sofia/gateway/musimi.dk/$1"/> </condition> </extension> and 9000_recordings.xml <extension name="Recordings"> <condition field="destination_number" expression="^\*(732673)$"> <action application="javascript" data="recordings.js"/> </condition> </extension> Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org ( /mc/compose?to=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
What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark The extensions are located under directory/default, and they look like this: <include> <user id="1001" mailbox="1001"> <params> <param name="password" value="1001"/> <param name="vm-password" value="1001"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="kenneth@fribert.dk"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Fribert"/> <variable name="effective_caller_id_number" value="12345678"/> </variables> </user> </include> <include> <user id="1002" mailbox="1002"> <params> <param name="password" value="1002"/> <param name="vm-password" value="1002"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="lian@fribert.dk"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1002"/> <variable name="user_context" value="default"/> </variables> </user> </include> As I understand them, the context set there, is the right one?
10-12-2009 kl. 00:00 skrev Mark Crane <mctch@yahoo.com> i meddelelsen <110796.60596.qm@web56401.mail.re3.yahoo.com>:
Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): <?xml version="1.0" encoding="utf-8"?> <!-- NOTICE: This context is usually accessed via authenticated callers on the sip profile on port 5060 or transfered callers from the public context which arrived via the sip profile on port 5080. Authenticated users will use the user_context variable on the user to determine what context they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml will build the domains acl using this value. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Example of doing things based on time of day. --> <extension name="tod_example" continue="true"> <!-- man strftime - M-F, 9AM to 6PM --> <condition field="${strftime(%w)}" expression="^([1-5])$"/> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="set" data="open=true"/> </condition> </extension> <extension name="global-intercept"> <condition field="destination_number" expression="^\*886$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="group-intercept"> <condition field="destination_number" expression="^\*8$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^\*870$"> <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/> </condition> </extension> <extension name="global" continue="true"> <condition field="${network_addr}" expression="^$" break="never"> <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/> <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/> </condition> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> <!-- This is an example of how to auto detect if telephone-event is missing and activate inband detection --> <!-- <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> --> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> </condition> </extension> <!-- If sip_req_host is not a local domain then this has to be an external sip uri --> <!-- <extension name="external_sip_uri" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="${outside_call}" expression="^$"/> <condition field="${domain_exists(${sip_req_host})}" expression="true"> <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/> </condition> </extension> --> <!-- snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference --> <extension name="snom-demo-2"> <condition field="destination_number" expression="^\*9001$"> <action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/> <action application="transfer" data="3000"/> </condition> </extension> <extension name="snom-demo-1"> <condition field="destination_number" expression="^\*9000$"> <!--<key> <light> <label> <user> <host> <profile> <action_name> <action>--> <action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*779$"> <action application="answer"/> <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/> <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/> <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/> <action application="eavesdrop" data="all"/> </condition> </extension> <extension name="call_return"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/> </condition> </extension> <extension name="del-group"> <condition field="destination_number" expression="^\*80(\d{2})$"> <action application="answer"/> <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 320)"/> </condition> </extension> <extension name="add-group"> <condition field="destination_number" expression="^\*81(\d{2})$"> <action application="answer"/> <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 640)"/> </condition> </extension> <extension name="call-group-simo"> <condition field="destination_number" expression="^\*82(\d{2})$"> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/> </condition> </extension> <extension name="call-group-order"> <condition field="destination_number" expression="^\*83(\d{2})$"> <action application="set" data="call_timeout=10"/> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="^\*8(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> <!-- dial the extension 3, 4, or 5 digit extensions for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="group_dial_sales"> <condition field="destination_number" expression="^\*2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/> </condition> </extension> <extension name="group_dial_support"> <condition field="destination_number" expression="^\*2001$"> <action application="bridge" data="group/support@${domain_name}"/> </condition> </extension> <extension name="group_dial_billing"> <condition field="destination_number" expression="^\*2002$"> <action application="bridge" data="group/billing@${domain_name}"/> </condition> </extension> <!-- voicemail operator extension --> <!-- <extension name="operator"> <condition field="destination_number" expression="^\*operator$|^0$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="transfer" data="1000 XML features"/> </condition> </extension> --> <!-- voicemail main2 extension --> <extension name="vmain2"> <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> <extension name="send_to_voicemail_5digits"> <condition field="destination_number" expression="^\*99(\d{5})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_4digits"> <condition field="destination_number" expression="^\*99(\d{4})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_3digits"> <condition field="destination_number" expression="^\*99(\d{3})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> <!-- start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml --> <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^\*(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/> </condition> </extension> <extension name="uwb_conferences"> <condition field="destination_number" expression="^\*(32\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@ultrawideband"/> </condition> </extension> <!-- MONO 48kHz conferences --> <extension name="cdquality_conferences"> <condition field="destination_number" expression="^\*(33\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@cdquality"/> </condition> </extension> <!-- dial the freeswitch conference via SIP--> <extension name="freeswitch_public_conf_via_sip"> <condition field="destination_number" expression="^\*9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> <!-- This will take the SAS from the b-leg and send it to the display on the a-leg phone. Known working with Polycom and Snom maybe others. --> <!-- <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/> <action application="export" data="nolocal:zrtp_secure_media=true"/> --> <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/> </condition> </extension> <!-- This extension will start a conference and invite a group. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/> </condition> </extension> <!-- This extension will start a conference and invite a few of people. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0912$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/> <action application="set" data="conference_auto_outcall_caller_id_number=0912"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/> </condition> </extension> <!--This extension will start a conference and invite several people upon entering --> <extension name="mad_boss"> <condition field="destination_number" expression="^\*0913$"> <!--These params effect the outcalls made once you join--> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>--> <!--Add as many of these as you need, These are the people you are going to call--> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss3@default"/> </condition> </extension> <!-- a sample IVR --> <extension name="ivr_demo"> <condition field="destination_number" expression="^\*5000$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <!-- Create a conference on the fly and pull someone in at the same time. --> <extension name="dynamic_conference"> <condition field="destination_number" expression="^\*5001$"> <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> </condition> </extension> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^\*pagegroup$|^\*7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> <!-- Parking extensions... transferring calls to 5900 will park them in a queue. --> <extension name="park"> <condition field="destination_number" expression="^\*5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name} in"/> </condition> </extension> <!-- Parking pickup extension. Calling 5901 will pickup the call. --> <extension name="unpark"> <condition field="destination_number" expression="^\*5901$"> <action application="answer"/> <action application="fifo" data="5900@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with snom phones. Set a function key to park+lot (lot being a number or name.) Set type to Park+Orbit. You can then park and pickup using the softkey on the phone. Should work with other phones. --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="park\+(\d+)"> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- The extension is parking pickup with a to param of the fifo we are calling Some phones send things like orbit= and you can extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^parking$"/> <condition field="${sip_to_params}" expression="fifo\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with linksys phones. Set a Phone tab option Call Park Serv to yes. You can park and pickup using soft keys "park" and "unpark" found during active call when moving navigation button. The other option is to use phone's star codes (defaults to *38 and *39). --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="callpark"/> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- This extension is used with linksys phones. The extension is parking pickup with a to param of the fifo we are calling. Linksys sends orbit=<parkingslotnumber> and we extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="pickup"/> <condition field="${sip_to_params}" expression="orbit\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- Here are some examples of how to override the ringback heard by the far end. You have two variables that you can use to override this. ringback - used when a call isn't answered. (early media) transfer_ringback - used when the call is already answered. (post answer) --> <!-- Demonstration of how to override the ringback in various situations --> <extension name="wait"> <condition field="destination_number" expression="^wait$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^\*9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^\*9979$"> <action application="txfax" data="/tmp/txfax.tif"/> <action application="hangup"/> </condition> </extension> <!-- Send a 180 and let the far end generate ringback. --> <extension name="ringback_180"> <condition field="destination_number" expression="^\*9980$"> <action application="ring_ready"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <!-- Send a 183 and send uk-ring as the ringtone. (early media) --> <extension name="ringback_183_uk_ring"> <condition field="destination_number" expression="^\*9981$"> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Send a 183 and use music as the ringtone. (early media) --> <extension name="ringback_183_music_ring"> <condition field="destination_number" expression="^\*9982$"> <action application="set" data="ringback=$${hold_music}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_uk_ring"> <condition field="destination_number" expression="^\*9983$"> <action application="set" data="transfer_ringback=$${uk-ring}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_music"> <condition field="destination_number" expression="^\*9984$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <extension name="ClueCon"> <condition field="destination_number" expression="^\*9991$"> <action application="set" data="effective_caller_id_name=ClueCon"/> <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/> </condition> </extension> <extension name="show_info"> <condition field="destination_number" expression="^\*9992$"> <action application="answer"/> <action application="info"/> <action application="sleep" data="250"/> <action application="hangup"/> </condition> </extension> <extension name="video_record"> <condition field="destination_number" expression="^\*9993$"> <action application="answer"/> <action application="record_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="video_playback"> <condition field="destination_number" expression="^\*9994$"> <action application="answer"/> <action application="play_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="delay_echo"> <condition field="destination_number" expression="^\*9995$"> <action application="answer"/> <action application="delay_echo" data="5000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^\*9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="milliwatt"> <condition field="destination_number" expression="^\*9997$"> <action application="answer"/> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/> </condition> </extension> <extension name="tone_stream"> <condition field="destination_number" expression="^\*9998$"> <action application="answer"/> <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> </condition> </extension> <!-- You will no longer hear the bong tone. The wav file is playing stating the call is secure. The file will not play unless you have both TLS and SRTP active. --> <extension name="hold_music"> <condition field="destination_number" expression="^\*9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <!-- This really should be an IVR for zrtp enrollment but this is just a demo--> <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> <!-- You can place files in the default directory to get included. --> <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>--> <!-- WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING Anything you put below this line will usually get ignored due to the file in default/99999_enum.xml as it will transfer the call to the enum dialplan. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <extension name="enum"> <condition field="${module_exists(mod_enum)}" expression="true"/> <condition field="destination_number" expression="^(.*)$"> <action application="transfer" data="$1 enum"/> </condition> </extension> <!-- <extension name="refer"> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression> </condition> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:(.*)@(.*)>]]></expression> <action application="set" data="refer_user=$1"/> <action application="set" data="refer_domain=$2"/> <action application="info"/> <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/> </condition> </extension> --> <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <!-- <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^\*7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/> <action application="execute_extension" data="ruri"/> </condition> </extension> --> <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! --> </context> </include> Then I have under default dir: musimidk.xml <extension name="musimi.dk"> <condition field="^0(.\d+)$" expression=""> <action application="bridge" data="sofia/gateway/musimi.dk/$1"/> </condition> </extension> and 9000_recordings.xml <extension name="Recordings"> <condition field="destination_number" expression="^\*(732673)$"> <action application="javascript" data="recordings.js"/> </condition> </extension> Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ 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 -----Inline Attachment Follows----- _______________________________________________ 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
Hi Mark et al. Don't know where my reply went, but certainly not to the list :-) So my public: default.xml looks like this; <!-- NOTICE: This context is usually accessed via the external sip profile sitting on port 5080. It is recommended to have separate inbound and outbound contexts. Not only for security but clearing up why you would need to do such a thing. You don't want outside un-authenticated callers hitting your default context which allows dialing calls thru your providers and results in Toll Fraud. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Tag anything pass thru here as an outside_call so you can make sure not to create any routing loops based on the conditions that it came from the outside of the switch. --> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> </condition> </extension> <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> </extension> <!-- <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> --> <!-- You can place files in the public directory to get included. --> <X-PRE-PROCESS cmd="include" data="public/*.xml"/> <!-- If you have made it this far lets challenge the caller and if they authenticate lets try what they dialed in the default context. (commented out by default) --> <!-- <extension name="check_auth" continue="true"> <condition field="${sip_authorized}" expression="^true$" break="never"> <anti-action application="respond" data="407"/> </condition> </extension> <extension name="transfer_to_default"> <condition> <action application="transfer" data="${destination_number} XML default"/> </condition> </extension> --> </context> </include> If I don't have any phones externally to my FS, could I just empty the public's default.xml? BR Fribert
11-12-2009 kl. 10:44 skrev Mark Crane <mctch@yahoo.com> i meddelelsen <117186.49301.qm@web56406.mail.re3.yahoo.com>:
What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark The extensions are located under directory/default, and they look like this: <include> <user id="1001" mailbox="1001"> <params> <param name="password" value="1001"/> <param name="vm-password" value="1001"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="kenneth@fribert.dk"/ ( /mc/compose?to=kenneth@fribert.dk%22/ )> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Fribert"/> <variable name="effective_caller_id_number" value="12345678"/> </variables> </user> </include> <include> <user id="1002" mailbox="1002"> <params> <param name="password" value="1002"/> <param name="vm-password" value="1002"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-keep-local-after-email" value="true"/> <param name="vm-mailto" value="lian@fribert.dk"/ ( /mc/compose?to=lian@fribert.dk%22/ )> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1002"/> <variable name="user_context" value="default"/> </variables> </user> </include> As I understand them, the context set there, is the right one?
10-12-2009 kl. 00:00 skrev Mark Crane <mctch@yahoo.com> i meddelelsen <110796.60596.qm@web56401.mail.re3.yahoo.com>:
Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist <mailinglist@fribert.dk> wrote: From: mailinglist <mailinglist@fribert.dk> Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): <?xml version="1.0" encoding="utf-8"?> <!-- NOTICE: This context is usually accessed via authenticated callers on the sip profile on port 5060 or transfered callers from the public context which arrived via the sip profile on port 5080. Authenticated users will use the user_context variable on the user to determine what context they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml will build the domains acl using this value. --> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <!-- Example of doing things based on time of day. --> <extension name="tod_example" continue="true"> <!-- man strftime - M-F, 9AM to 6PM --> <condition field="${strftime(%w)}" expression="^([1-5])$"/> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="set" data="open=true"/> </condition> </extension> <extension name="global-intercept"> <condition field="destination_number" expression="^\*886$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="group-intercept"> <condition field="destination_number" expression="^\*8$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^\*870$"> <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/> </condition> </extension> <extension name="global" continue="true"> <condition field="${network_addr}" expression="^$" break="never"> <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/> <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/> </condition> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> <!-- This is an example of how to auto detect if telephone-event is missing and activate inband detection --> <!-- <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> --> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition> <condition> <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/> </condition> </extension> <!-- If sip_req_host is not a local domain then this has to be an external sip uri --> <!-- <extension name="external_sip_uri" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="${outside_call}" expression="^$"/> <condition field="${domain_exists(${sip_req_host})}" expression="true"> <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/> </condition> </extension> --> <!-- snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference --> <extension name="snom-demo-2"> <condition field="destination_number" expression="^\*9001$"> <action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/> <action application="transfer" data="3000"/> </condition> </extension> <extension name="snom-demo-1"> <condition field="destination_number" expression="^\*9000$"> <!--<key> <light> <label> <user> <host> <profile> <action_name> <action>--> <action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/> </condition> </extension> <extension name="eavesdrop"> <condition field="destination_number" expression="^\*779$"> <action application="answer"/> <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/> <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/> <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/> <action application="eavesdrop" data="all"/> </condition> </extension> <extension name="call_return"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/> </condition> </extension> <extension name="del-group"> <condition field="destination_number" expression="^\*80(\d{2})$"> <action application="answer"/> <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 320)"/> </condition> </extension> <extension name="add-group"> <condition field="destination_number" expression="^\*81(\d{2})$"> <action application="answer"/> <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/> <action application="gentones" data="%(1000, 0, 640)"/> </condition> </extension> <extension name="call-group-simo"> <condition field="destination_number" expression="^\*82(\d{2})$"> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/> </condition> </extension> <extension name="call-group-order"> <condition field="destination_number" expression="^\*83(\d{2})$"> <action application="set" data="call_timeout=10"/> <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="^\*8(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> <!-- dial the extension 3, 4, or 5 digit extensions for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension"> <condition field="destination_number" expression="(^\d{5}$|^\d{4}$|^\d{3}$)"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="group_dial_sales"> <condition field="destination_number" expression="^\*2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/> </condition> </extension> <extension name="group_dial_support"> <condition field="destination_number" expression="^\*2001$"> <action application="bridge" data="group/support@${domain_name}"/> </condition> </extension> <extension name="group_dial_billing"> <condition field="destination_number" expression="^\*2002$"> <action application="bridge" data="group/billing@${domain_name}"/> </condition> </extension> <!-- voicemail operator extension --> <!-- <extension name="operator"> <condition field="destination_number" expression="^\*operator$|^0$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="transfer" data="1000 XML features"/> </condition> </extension> --> <!-- voicemail main2 extension --> <extension name="vmain2"> <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/> </condition> </extension> <extension name="send_to_voicemail_5digits"> <condition field="destination_number" expression="^\*99(\d{5})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_4digits"> <condition field="destination_number" expression="^\*99(\d{4})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="send_to_voicemail_3digits"> <condition field="destination_number" expression="^\*99(\d{3})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> <!-- dial via SIP uri --> <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> <!-- start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml --> <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^\*(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband"/> </condition> </extension> <extension name="uwb_conferences"> <condition field="destination_number" expression="^\*(32\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@ultrawideband"/> </condition> </extension> <!-- MONO 48kHz conferences --> <extension name="cdquality_conferences"> <condition field="destination_number" expression="^\*(33\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@cdquality"/> </condition> </extension> <!-- dial the freeswitch conference via SIP--> <extension name="freeswitch_public_conf_via_sip"> <condition field="destination_number" expression="^\*9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> <!-- This will take the SAS from the b-leg and send it to the display on the a-leg phone. Known working with Polycom and Snom maybe others. --> <!-- <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/> <action application="export" data="nolocal:zrtp_secure_media=true"/> --> <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/> </condition> </extension> <!-- This extension will start a conference and invite a group. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/> </condition> </extension> <!-- This extension will start a conference and invite a few of people. At anytime the participant can dial *2 to bridge directly to the boss. All other callers are then hung up on. --> <extension name="mad_boss_intercom"> <condition field="destination_number" expression="^\*0912$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/> <action application="set" data="conference_auto_outcall_caller_id_number=0912"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/> </condition> </extension> <!--This extension will start a conference and invite several people upon entering --> <extension name="mad_boss"> <condition field="destination_number" expression="^\*0913$"> <!--These params effect the outcalls made once you join--> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>--> <!--Add as many of these as you need, These are the people you are going to call--> <action application="conference_set_auto_outcall" data="loopback/9999"/> <action application="conference" data="madboss3@default"/> </condition> </extension> <!-- a sample IVR --> <extension name="ivr_demo"> <condition field="destination_number" expression="^\*5000$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <!-- Create a conference on the fly and pull someone in at the same time. --> <extension name="dynamic_conference"> <condition field="destination_number" expression="^\*5001$"> <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/> </condition> </extension> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^\*pagegroup$|^\*7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> <!-- Parking extensions... transferring calls to 5900 will park them in a queue. --> <extension name="park"> <condition field="destination_number" expression="^\*5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name} in"/> </condition> </extension> <!-- Parking pickup extension. Calling 5901 will pickup the call. --> <extension name="unpark"> <condition field="destination_number" expression="^\*5901$"> <action application="answer"/> <action application="fifo" data="5900@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with snom phones. Set a function key to park+lot (lot being a number or name.) Set type to Park+Orbit. You can then park and pickup using the softkey on the phone. Should work with other phones. --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="park\+(\d+)"> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- The extension is parking pickup with a to param of the fifo we are calling Some phones send things like orbit= and you can extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^parking$"/> <condition field="${sip_to_params}" expression="fifo\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- This extension is used with linksys phones. Set a Phone tab option Call Park Serv to yes. You can park and pickup using soft keys "park" and "unpark" found during active call when moving navigation button. The other option is to use phone's star codes (defaults to *38 and *39). --> <extension name="park"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="callpark"/> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression> <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/> </condition> </extension> <!-- This extension is used with linksys phones. The extension is parking pickup with a to param of the fifo we are calling. Linksys sends orbit=<parkingslotnumber> and we extract that info. --> <extension name="unpark"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="pickup"/> <condition field="${sip_to_params}" expression="orbit\=(\d+)"> <action application="answer"/> <action application="fifo" data="$1@${domain_name} out nowait"/> </condition> </extension> <!-- Here are some examples of how to override the ringback heard by the far end. You have two variables that you can use to override this. ringback - used when a call isn't answered. (early media) transfer_ringback - used when the call is already answered. (post answer) --> <!-- Demonstration of how to override the ringback in various situations --> <extension name="wait"> <condition field="destination_number" expression="^wait$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^\*9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^\*9979$"> <action application="txfax" data="/tmp/txfax.tif"/> <action application="hangup"/> </condition> </extension> <!-- Send a 180 and let the far end generate ringback. --> <extension name="ringback_180"> <condition field="destination_number" expression="^\*9980$"> <action application="ring_ready"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> <!-- Send a 183 and send uk-ring as the ringtone. (early media) --> <extension name="ringback_183_uk_ring"> <condition field="destination_number" expression="^\*9981$"> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Send a 183 and use music as the ringtone. (early media) --> <extension name="ringback_183_music_ring"> <condition field="destination_number" expression="^\*9982$"> <action application="set" data="ringback=$${hold_music}"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_uk_ring"> <condition field="destination_number" expression="^\*9983$"> <action application="set" data="transfer_ringback=$${uk-ring}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <!-- Answer the call and use music as the ringtone. (post answer) --> <extension name="ringback_post_answer_music"> <condition field="destination_number" expression="^\*9984$"> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="answer"/> <action application="bridge" data="loopback/wait"/> </condition> </extension> <extension name="ClueCon"> <condition field="destination_number" expression="^\*9991$"> <action application="set" data="effective_caller_id_name=ClueCon"/> <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/> </condition> </extension> <extension name="show_info"> <condition field="destination_number" expression="^\*9992$"> <action application="answer"/> <action application="info"/> <action application="sleep" data="250"/> <action application="hangup"/> </condition> </extension> <extension name="video_record"> <condition field="destination_number" expression="^\*9993$"> <action application="answer"/> <action application="record_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="video_playback"> <condition field="destination_number" expression="^\*9994$"> <action application="answer"/> <action application="play_fsv" data="/tmp/testrecord.fsv"/> </condition> </extension> <extension name="delay_echo"> <condition field="destination_number" expression="^\*9995$"> <action application="answer"/> <action application="delay_echo" data="5000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^\*9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="milliwatt"> <condition field="destination_number" expression="^\*9997$"> <action application="answer"/> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/> </condition> </extension> <extension name="tone_stream"> <condition field="destination_number" expression="^\*9998$"> <action application="answer"/> <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> </condition> </extension> <!-- You will no longer hear the bong tone. The wav file is playing stating the call is secure. The file will not play unless you have both TLS and SRTP active. --> <extension name="hold_music"> <condition field="destination_number" expression="^\*9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <!-- This really should be an IVR for zrtp enrollment but this is just a demo--> <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> <!-- You can place files in the default directory to get included. --> <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>--> <!-- WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING Anything you put below this line will usually get ignored due to the file in default/99999_enum.xml as it will transfer the call to the enum dialplan. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <extension name="enum"> <condition field="${module_exists(mod_enum)}" expression="true"/> <condition field="destination_number" expression="^(.*)$"> <action application="transfer" data="$1 enum"/> </condition> </extension> <!-- <extension name="refer"> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression> </condition> <condition field="${sip_refer_to}"> <expression><![CDATA[<sip:(.*)@(.*)>]]></expression> <action application="set" data="refer_user=$1"/> <action application="set" data="refer_domain=$2"/> <action application="info"/> <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/> </condition> </extension> --> <!-- This is an example of how to override the RURI on an outgoing invite to a registered contact. --> <!-- <extension name="ruri"> <condition field="destination_number" expression="^ruri$"> <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> <extension name="7004"> <condition field="destination_number" expression="^\*7004$"> <action application="set" data="ruri_profile=default"/> <action application="set" data="ruri_user=2000"/> <action application="set" data="ruri_contact=1001@${domain_name}"/> <action application="execute_extension" data="ruri"/> </condition> </extension> --> <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! --> </context> </include> Then I have under default dir: musimidk.xml <extension name="musimi.dk"> <condition field="^0(.\d+)$" expression=""> <action application="bridge" data="sofia/gateway/musimi.dk/$1"/> </condition> </extension> and 9000_recordings.xml <extension name="Recordings"> <condition field="destination_number" expression="^\*(732673)$"> <action application="javascript" data="recordings.js"/> </condition> </extension> Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ 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 -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org ( /mc/compose?to=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
Hello, I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and fifo_consumer_caller_import variables, but i don't know use it. I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/
I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro <dftoro@yahoo.com>
Hello,
I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and fifo_consumer_caller_import variables, but i don't know use it.
I appreciate any suggestion
Diego Toro http://lacarretade.blogspot.com/
_______________________________________________ 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
Thanks for your answer, I want to resolver the caller id to dialplan level, I know variables fifo_caller_consumer_import and fifo_consumer_caller_import but isn't clear for me how use it. Diego Toro http://lacarretade.blogspot.com/ --- On Sun, 12/13/09, Seven Du <dujinfang@gmail.com> wrote: From: Seven Du <dujinfang@gmail.com> Subject: Re: [Freeswitch-users] fifo caller id To: freeswitch-users@lists.freeswitch.org Date: Sunday, December 13, 2009, 7:54 AM I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro <dftoro@yahoo.com> Hello, I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and fifo_consumer_caller_import variables, but i don't know use it. I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ _______________________________________________ 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 -----Inline Attachment Follows----- _______________________________________________ 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
Hey There, I came in not seeing any former posts of yours, so if this one is unhelpful, just delete. I did my FS install using PFsense as well. Its been working famously for a few months now. Very happy. I wrote what I discovered for FS on PFsense in this wiki: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I assume since you're using PFsense that your computer is functioning as a firewall AND a phone system (a dual homed host, in other words). That's what the wiki attempt above is aimed at. If you follow those instructions, you'll send and receive calls (as we were and are able to). It is working for us, at least. One problem in your case: I didn't really like using the PFsense Web interface when configuring FS (except for installing FS and setting some system parameters. Its great for PFsense, though). It helped me more to get in with ssh and vi and make FS work. Having done that successfully, you'll be more likely to effectively use the PFsense web interface for FS, as it's really just a "short cut" for someone that understands the FS file system, in my opinion. Good luck, Mike G. On Tue, Dec 8, 2009 at 1:20 PM, mailinglist <mailinglist@fribert.dk> wrote:
Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml
Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it.
How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time.
08-12-2009 kl. 07:05 skrev "mailinglist" <mailinglist@fribert.dk> i meddelelsen <4B1DFABC020000E1000002C2@mail.fribert.dk>: Hi Mark
Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error.
I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch.
Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number:
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( sofia/internal/1001@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan?
Best regards Kenneth
08-12-2009 kl. 03:05 skrev Mark Crane <mctch@yahoo.com> i meddelelsen 659603.29094.qm@web56408.mail.re3.yahoo.com:
Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch@firewall.fribert.dk<http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch@firewall.fribert.dk>> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory)
I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary?
Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier.
Question ----------------------------------------------
Extension Name musimi.dk Enabled true Order 001 Description ...
condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1
Response: ----------------------------------------------
This is correct as long as you have a gateway that is registered called musimi.dk
Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ...
action bridge sofia/internal/$
Response: ----------------------------------------------
No idea what this is for its not needed as far as I can tell.
Now please summarize what you still need help on.
Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer
_______________________________________________ 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
participants (6)
-
Diego Toro
-
mailinglist
-
Mark Crane
-
Michael Gende
-
Nandy Dagondon
-
Seven Du