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digit_action_set_realm question
by Abaci B 06 Jan '16

06 Jan '16

Hi, I was wondering if there is any way to unset the bind_digit_action realm once it's set using digit_action_set_realm, I tried setting it to "_UNDEF_" and "" none which seem to work.

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Authentication of REGISTERs through a RADIUS server
by Daniel Corbe 05 Jan '16

05 Jan '16

Hello, I took note of mod_xml_radius but the documentation on this module is thin in comparison to the rest of the mod set. It isn’t clear to me how I’d tell FreeSWITCH to begin routing REGISTER requests to my RADIUS server once I’ve enabled and configured the mod. I’ve got a pretty simple configuration in place as I’m only interested in registrations (below). Is there something I need to do now either from the dialplan or the Sofia config? Someone hitting me with a cluebat here would be lovely. <configuration name="xml_radius.conf" description="Radius XML Gateway"> <auth_reg> <connection name="testing"> <param name="authserver" value="192.168.164.137:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="request-type=user"/> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param name="User-Name" variable="sip_from_user" format="%s"/> <param name="Digest-Response" variable="sip_auth_response" format="%s"/> <param name="Digest-Realm" variable="sip_auth_realm" format="%s"/> <param name="Digest-Nonce" variable="sip_auth_nonce" format="%s"/> <param name="Digest-Username" variable="sip_auth_username" format="%s"/> <param name="Digest-URI" variable="sip_auth_uri" format="%s"/> <param name="Digest-Method" variable="sip_auth_method" format="%s"/> <param name="Digest-Algorithm" variable="sip_auth_method" format="MD5"/> <param name="Digest-Qop" variable="sip_auth_qop" format="%s"/> <param name="Digest-CNonce" variable="sip_auth_cnonce" format="%s"/> <param name="Digest-Nonce-Count" variable="sip_auth_nc" format="%s"/> </fields> </auth_reg> <configuration> freeswitch(a)voice-lab.hammerfiber.net> load mod_xml_radius 2016-01-05 12:43:33.179332 [INFO] mod_xml_radius.c:170 Could not find 'auth_invite' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_enum.c:880 ENUM Reloaded 2016-01-05 12:43:33.198334 [INFO] switch_time.c:1415 Timezone reloaded 1781 definitions 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:257 Could not find 'auth_app' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:291 Could not find 'acct_start' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:325 Could not find 'acct_end' section in config file. 2016-01-05 12:43:33.198334 [CONSOLE] switch_loadable_module.c:1538 Successfully Loaded [mod_xml_radius] 2016-01-05 12:43:33.198334 [NOTICE] switch_loadable_module.c:292 Adding Application 'radius_auth' 2016-01-05 12:43:33.198334 [NOTICE] switch_loadable_module.c:338 Adding API Function ‘xml_radius_debug'

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outbound proxy and port
by Chris B. Ware 05 Jan '16

05 Jan '16

Hi, I need to send registrations and calls from external profile to a si provider, passing through an outbound proxy, port 5080. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151"/> Registrations pass through the OB Proxy, but using, of course, standard 5060 port. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151:5080"/> Param seems to be not understood and registrations go directly to the sip provider. How can I set outbound proxy port? Thanks

7 7

Call rejecting with asterisk
by Rutu Patel 05 Jan '16

05 Jan '16

Hi All, I am trying to send calls from freeswitch to asterisk. I have setup gateway in freeswitch with proxy as asterisk IP. But when I dial the number it is sending the call to asterisk and showing as below in asterisk: username mismatch <123456> , digest has <FreeSWITCH> and then call rejected. I have tried by setting From-User = 123456 in gateway but no luck. 123456 is an extension on freeswitch. Can anyone please help to solve this ? -- Thanks, Rutu Patel <http://www.inextrix.com>

1 0

mod_verto: WS SETUP FAILED [�M 5g]
by Vitaly Kovalyshyn 05 Jan '16

05 Jan '16

Hello! FreeSWITCH (Version 1.6.5 git d5520a6 2015-11-19 20:27:21Z 64bit) mod_verto with WSS: None stop in the FreeSWITCH log: 2016-01-05 06:52:19.654908 [INFO] mod_verto.c:1973 194.183.YY.XX:60874 Starting client thread. 2016-01-05 06:52:19.674904 [INFO] mod_verto.c:3997 194.183.YY.XX:60873 Client Connect. 2016-01-05 06:52:19.794942 [DEBUG] mod_verto.c:1820 194.183.YY.XX:57780 WS SETUP FAILED [_��] 2016-01-05 06:52:19.834964 [INFO] mod_verto.c:2000 194.183.YY.XX:57780 Ending client thread. 2016-01-05 06:52:19.834964 [INFO] mod_verto.c:2007 194.183.YY.XX:57780 Thread ended 2016-01-05 07:00:32.004805 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15759 WS SETUP FAILED [`?)g] 2016-01-05 07:00:32.004805 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15771 WS SETUP FAILED ] 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2000 194.183.YY.XX:15759 Ending client thread. 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2000 194.183.YY.XX:15771 Ending client thread. 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2007 194.183.YY.XX:15759 Thread ended 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2007 194.183.YY.XX:15771 Thread ended 2016-01-05 07:00:32.004805 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15770 WS SETUP FAILED [1g] 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2000 194.183.YY.XX:15770 Ending client thread. 2016-01-05 07:00:32.004805 [INFO] mod_verto.c:2007 194.183.YY.XX:15770 Thread ended 2016-01-05 07:00:32.044808 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15746 WS SETUP FAILED ] 2016-01-05 07:00:32.044808 [INFO] mod_verto.c:2000 194.183.YY.XX:15746 Ending client thread. 2016-01-05 07:00:32.044808 [INFO] mod_verto.c:2007 194.183.YY.XX:15746 Thread ended 2016-01-05 07:00:32.064935 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15757 WS SETUP FAILED [�M 5g] 2016-01-05 07:00:32.064935 [INFO] mod_verto.c:2000 194.183.YY.XX:15757 Ending client thread. 2016-01-05 07:00:32.064935 [INFO] mod_verto.c:2007 194.183.YY.XX:15757 Thread ended 2016-01-05 07:00:32.064935 [DEBUG] mod_verto.c:1820 194.183.YY.XX:15768 WS SETUP FAILED ] And FreeSWITCH use all free memory and swap in Debian Linux 8.2. What does it mean ? Best regards, Vitaly Kovalyshyn

3 4

FreeSWITCHers' 2015 has been GREAT
by Giovanni Maruzzelli 04 Jan '16

04 Jan '16

Heya fellas, first of all I want to thank our Supreme Master, Anthony Minessale II, and the Jedi Knights Brian West, Mike Jerris, Ken Rice, William King, Kathleen King, Italo Rossi. It's because of these persons, of their craft, of their sleepless nights, of their creativity, vision and dedication that we can use and contribute to the most advanced audio/video application platform in the known Universe. Then I want to thank our beautiful, friendly, caring and wise Community. We're never alone. Forward! Toward further future achievements ! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Call rejecting while IP in the ACL
by Muhammad Naseer Bhatti 04 Jan '16

04 Jan '16

I am trying to authenticate with cidr attribute for the directory and configure sofia profile is to use apply-inbound-acl for value: domains. While the IP address is set there in the ACL list and seen by the ACL, the call is not hitting the dialplan. freeswitch@internal> acl 10.211.55.2 10.211.55.0/24 true freeswitch@internal> reloadacl +OK acl reloaded 2016-01-03 16:05:14.920901 [NOTICE] switch_core.c:1568 Created ip list domains default (deny) 2016-01-03 16:05:14.920901 [NOTICE] switch_utils.c:545 Adding 10.211.55.0/24 (allow) [8449347(a)10.211.55.26] to list domains 2016-01-03 16:05:31.141849 [NOTICE] switch_channel.c:1091 New Channel sofia/vBilling/8449347(a)10.211.55.26 [aaf18f0c-b21a-11e5-8b74-3f5672d9ef1e] But when I make a call, it goes straight to HANGUP. 2016-01-03 16:08:35.941031 [NOTICE] switch_channel.c:1091 New Channel sofia/vBilling/8449347(a)10.211.55.26 [19164d10-b21b-11e5-8b77-3f5672d9ef1e] 2016-01-03 16:08:35.941031 [DEBUG] switch_core_state_machine.c:473 (sofia/vBilling/8449347(a)10.211.55.26) Running State Change CS_NEW 2016-01-03 16:08:35.941031 [DEBUG] sofia.c:9244 sofia/vBilling/8449347(a)10.211.55.26 receiving invite from 10.211.55.2:51025 version: 1.7.0 git de5bbef 2015-12-31 16:03:11Z 64bit 2016-01-03 16:08:35.941031 [DEBUG] switch_core_state_machine.c:492 (sofia/vBilling/8449347(a)10.211.55.26) State NEW 2016-01-03 16:08:35.941031 [DEBUG] sofia.c:2147 detaching session 19164d10-b21b-11e5-8b77-3f5672d9ef1e 2016-01-03 16:08:35.941031 [DEBUG] sofia.c:2255 Re-attaching to session 19164d10-b21b-11e5-8b77-3f5672d9ef1e 2016-01-03 16:08:35.960967 [DEBUG] sofia.c:9244 sofia/vBilling/8449347(a)10.211.55.26 receiving invite from 10.211.55.2:51025 version: 1.7.0 git de5bbef 2015-12-31 16:03:11Z 64bit 2016-01-03 16:08:35.960967 [NOTICE] sofia.c:2145 Hangup sofia/vBilling/8449347(a)10.211.55.26 [CS_NEW] [CALL_REJECTED] 2016-01-03 16:08:35.980992 [DEBUG] sofia.c:1355 Channel is already hungup. 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:473 (sofia/vBilling/8449347(a)10.211.55.26) Running State Change CS_HANGUP 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:739 (sofia/vBilling/8449347(a)10.211.55.26) Callstate Change DOWN -> HANGUP 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:741 (sofia/vBilling/8449347(a)10.211.55.26) State HANGUP 2016-01-03 16:08:35.980992 [DEBUG] mod_sofia.c:431 Channel sofia/vBilling/8449347(a)10.211.55.26 hanging up, cause: CALL_REJECTED 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:60 sofia/vBilling/8449347(a)10.211.55.26 Standard HANGUP, cause: CALL_REJECTED 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:741 (sofia/vBilling/8449347(a)10.211.55.26) State HANGUP going to sleep 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:508 (sofia/vBilling/8449347(a)10.211.55.26) State Change CS_HANGUP -> CS_REPORTING 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:473 (sofia/vBilling/8449347(a)10.211.55.26) Running State Change CS_REPORTING 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:827 (sofia/vBilling/8449347(a)10.211.55.26) State REPORTING 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:104 sofia/vBilling/8449347(a)10.211.55.26 Standard REPORTING, cause: CALL_REJECTED 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:827 (sofia/vBilling/8449347(a)10.211.55.26) State REPORTING going to sleep 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:499 (sofia/vBilling/8449347(a)10.211.55.26) State Change CS_REPORTING -> CS_DESTROY 2016-01-03 16:08:35.980992 [DEBUG] switch_core_session.c:1646 Session 5 (sofia/vBilling/8449347(a)10.211.55.26) Locked, Waiting on external entities 2016-01-03 16:08:35.980992 [NOTICE] switch_core_session.c:1664 Session 5 (sofia/vBilling/8449347(a)10.211.55.26) Ended 2016-01-03 16:08:35.980992 [NOTICE] switch_core_session.c:1668 Close Channel sofia/vBilling/8449347(a)10.211.55.26 [CS_DESTROY] 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:630 (sofia/vBilling/8449347(a)10.211.55.26) Running State Change CS_DESTROY 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:640 (sofia/vBilling/8449347(a)10.211.55.26) State DESTROY 2016-01-03 16:08:35.980992 [DEBUG] mod_sofia.c:341 sofia/vBilling/8449347(a)10.211.55.26 SOFIA DESTROY 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:111 sofia/vBilling/8449347(a)10.211.55.26 Standard DESTROY 2016-01-03 16:08:35.980992 [DEBUG] switch_core_state_machine.c:640 (sofia/vBilling/8449347(a)10.211.55.26) State DESTROY going to sleep Can’t seem to figure out if I am missing something here. My acl is served as, Thanks, Muhammad

3 8

sip simple delivery
by Juan Pablo L. 04 Jan '16

04 Jan '16

Hi, i would like to find out how can i stop a sip simple from being delivered once it has been dealt with, i have a module that receives the event and decides whether to deliver it or not based on some criteria ($$), but it does not matter what the module does, after it has finished precessing the message, freeswitch still delivers it. i understand it does this by default, any advise would be appreciated. thanks!

5 8

Alsa to RTP
by Pete Kay 04 Jan '16

04 Jan '16

Hi I am able to use aplay to playback media from a wav file and arecord to record from sound card to a wav file. I would like to playback the media to freeswitch via playback and send the RTP to arecord. Does anyone know how to do it? Is there any sample that shows how this can be done? Thanks, Pete

2 1

Add codec
by Bhavik Patel 04 Jan '16

04 Jan '16

Hi, in FreeSWITCH, how can I add following codec: codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE -- Bhavik Patel DA-IICT

3 7