- FreeSWITCH-users - FreeSWITCH Mailing Lists

web voice mail interface / web-vm.tpl
by John Wehle 05 Aug '08

05 Aug '08

I'm looking for a web interface to listen to / delete voicemail messages. I see web-vm.tpl, however I'm not sure what to do with it. How do I go about getting a page which prompts for user id / password, logs in, displays a list of messages, and allows for play back / deleting a message? I'm not adverse to doing some scripting / web page development, however I'd prefer a pointer in the right direction and I'm also not looking to reinvent the wheel. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] How to compile and load mod_cdr
by Shehzad Pankhawala 05 Aug '08

05 Aug '08

Hi, ok, Mike, i try to do that, meanwhile, dear Anthony, you please the same you told if you have contact with author and then reply if any interesting come out.

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IPv6 capable?
by Rick Giliberti 05 Aug '08

05 Aug '08

I would rather avoid using NAT traversal techniques and prefer to use IPv6 to get a public address. Does FreeSWITCH support IPv6 on 32-bit Windows? If not, is there a timeframe when it may be implemented?

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Woomera - Segmentation fault
by Shehzad Pankhawala 05 Aug '08

05 Aug '08

I need to configure H323 with freeswitch. So, I Installed and configured as the wiki link: http://wiki.freeswitch.org/wiki/FreeSwitch_H323 Then i registerd one phone with H323 gateway, given Gateway ip of my freeswitch. eg. with number 5678 And second phone is registered normaly. eg. with number 1001 When i call from 5678 to 1001 it works fine, after hangup the woomera daemon shutdown with error displaying: ---------------------------------------------------- Incoming connection from 127.0.0.1 0:11.917 WoomeraThread:b7d04cb8 connection.cxx(57) Woomera Sending command EVENT HELLO 0 1.0 Post Increment 0:11.917 WoomeraThread:b7d04cb8 connection.cxx(30) Woomera Sending response 200 Listener enabled 0:11.918 WoomeraThread:b7d04cb8 connection.cxx(57) Woomera Sending command EVENT INCOMING 1 0:11.927 WoomeraThread:b7d04cb8 connection.cxx(57) Woomera Sending command EVENT MEDIA 1 AUDIO 0:11.929 WoomeraThread:b7d04cb8 connection.cxx(30) Woomera Sending response 200 Call answered 0:16.651 H323 Cleaner h323.cxx(895) H323 Connection ip$192.168.1.23:56984/4096 terminated. 0:16.652 H323 Cleaner openh323.cxx(877) Woomera OnCleared received 0:16.652 H323 Cleaner openh323.cxx(563) Shutting down non-transferred call 0:16.653 H323 Cleaner connection.cxx(57) Woomera Sending command EVENT HANGUP 1 0:16.655 WoomeraThread:b7d04cb8 connection.cxx(30) Woomera Sending response 100 Clearing call Segmentation fault ---------------------------------------------------------- In freeswitch console the following comes: ---------------------------------------------------------------- 2008-08-05 20:30:36 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes [cd426778-2871-4903-b44f-698668f86ff9] 2008-08-05 20:30:36 [NOTICE] sofia.c:2172 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes! 2008-08-05 20:30:38 [NOTICE] sofia.c:2496 sofia_handle_sip_i_state() Channel [sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes] has been answered 2008-08-05 20:30:41 [NOTICE] mod_woomera.c:895 woomera_channel_thread_run() Hangup Woomera/1009-a996 [CS_EXECUTE] [NORMAL_CLEARING] 2008-08-05 20:30:41 [ERR] switch_core_io.c:746 switch_core_session_write_frame() Write Buffer 0 bytes Failed! 2008-08-05 20:30:41 [NOTICE] switch_ivr_bridge.c:382 audio_bridge_on_exchange_media() Hangup sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes [CS_RESET] [NORMAL_CLEARING] 2008-08-05 20:30:41 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 84 (sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes) Ended 2008-08-05 20:30:41 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1009@192.168.1.23:5061;rinstance=da8ec472e5aec95d;fs_nat=yes [CS_HANGUP] 2008-08-05 20:30:41 [ALERT] mod_woomera.c:1089 woomera_thread_run() {main} HELP! I lost my connection to woomera! 2008-08-05 20:30:41 [WARNING] mod_woomera.c:463 tech_destroy() {main} Already Disconnected 2008-08-05 20:30:41 [WARNING] mod_woomera.c:777 woomera_locate_socket() {main} Cannot Reconnect to Woomera! retry in 5 seconds 2008-08-05 20:30:41 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 83 (Woomera/1009-a996) Ended 2008-08-05 20:30:41 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel Woomera/1009-a996 [CS_HANGUP] _______________________________________________________________________________________________________ Has anybody before faced with this problem.? Please reply. Thanks Shehzad

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Trying to call a remote extension at a remote FS
by Erol Akarsu 05 Aug '08

05 Aug '08

I am trying to call a remote extension '1000' at a remote FS and bridge with a local extension '1238'. I am having problem ' No Matching gateway found'. It will be highly appreciated if you can help us freeswitch@trvoip> originate {ignore_early_media=true,originate_timeout=60}sofia/default/sip:1000@192.168.1.5 1238 2008-08-05 13:34:24 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/internal/sip:1000@192.168.1.5 [befd6852-6314-11dd-8320-d73ed16c8417] 2008-08-05 13:34:24 [ERR] sofia_reg.c:915 sofia_reg_handle_sip_r_challenge() No Matching gateway found 2008-08-05 13:34:24 [NOTICE] sofia_reg.c:935 sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/sip:1000@192.168.1.5 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] API CALL [originate({ignore_early_media=true,originate_timeout=60}sofia/default/sip:1000@192.168.1.5 1238)] output: -ERR MANDATORY_IE_MISSING Thanks Erol Akarsu

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Forked dialing and CDRs
by Rupa Schomaker (lists) 05 Aug '08

05 Aug '08

I'm using XML CDRs, posting to a rails app to capture and display CDR records. Incoming calls are forked to both internal extensions and external cell phones. The mapping is looked up in the DB via a javascript call which hands back a dialstring similar to: 2008-08-05 10:09:21 [INFO] mod_dptools.c:862 log_function() dial_string={ignore_early_media=true}sofia/internal/101%192.168.1.20,sofia/internal/102%192.168.1.20,sofia/loopback/CELL%192.168.1.20:5061 The sofia/internal/##%domain works as expected. I get one CDR with everything stacked up in the XML. The sofia/loopback/##@domain:5061 generates a new CDR. This is expected but is a pain in the rear since it means that I have to cross-correlated log entries to show the one that one the race and suppress/roll up the rest. Ok, so why don't I do: sofia/gateway/mygateway/cellnumber%domain ? Because I want to do other work before dialing out the gateway and to centralize the work done for that dialout. Specifically, I do real-time rating to determine which actual gateway to route the call through and also handle failover if a particular gateway doesn't terminate the call with an expected error code. There is a level of abstraction missing here I think. The transfer application does exactly what I want to do. I give it a context and a # and it routes the call as expected. But transfer can't handle forked dialing. For that I need bridge. The bridge application can fork but requires too much knowledge about the transport/gateway/etc. Thoughts?

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C# EventSocket IVR Example?
by Gerry Hull 05 Aug '08

05 Aug '08

Does anyone have one? Gerry

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Inter-domain connection
by Alois Komenda 05 Aug '08

05 Aug '08

Hello, how do I realize an inter-domain connection with FreeSWITCH? I have two instances of FS. One in domain A and one in domain B. How can users in A call users in B (and vice versa)? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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recording in mono
by Ilan Perez 05 Aug '08

05 Aug '08

What setting do I change to change the record_session code to record in mono I have edited this line and it doesn't seem to do much. from <action application="set" data="RECORD_STEREO=true"/> To <action application="set" data="RECORD_STEREO=false"/> Anyone. Ilan Perez

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Re: [Freeswitch-users] inbound calls
by Roberto Hernandez 05 Aug '08

05 Aug '08

Figured it out. Thanks, Roberto _____ From: Roberto Hernandez [mailto:rh@bluevisor.com] Sent: Monday, August 04, 2008 11:01 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: inbound calls I've configured one X-Lite softphone (id=1000) and made my first few PSTN outbound calls with no problem. Inbound is a bit of a mystery. How do I configure PSTN inbound calls to terminated on my softphone? I found the following archived email, would someone break it down so new users can understand? Thanks in advance. In public.xml, you want something like this: <extension name="test_did"> <condition field="destination_number" expression="^(4153084258)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> and then in default.xml have something like this in the default context: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4153084258)$" continue="on-true"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_e xt}"/> </condition> </extension> Thanks, Roberto

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