December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Best way to run originate calls through dial plan
by eaf 03 Dec '09

03 Dec '09

What would be the best way of making originate() run call through a dial plan (compared to directly going to a specified VOIP gateway). Would it be loopbacks, i.e. smth like this? /opt/freeswitch/bin/fs_cli -x "originate {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js zzzzzzzzzz)'" The idea of this is that originate() sets up the first call, then webcall.js plays back a WAV, and bridges the first call with the second one (also set up via loopback). Thanks! -- View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Remote fetching of voicemail
by François Legal 03 Dec '09

03 Dec '09

Thanks. I did not succed to fincing the correct syntx with inline, but the transfer application did work. François On Wed, 2 Dec 2009 12:21:54 -0600, Anthony Minessale wrote: bind to the transfer app so that it transfers the call to the vm extension that way the current application is always interrupted and replaced. The special "inline" dialplan lets you transfer calls right to an application use "inline" as the dp name and voicemail: as the extension On Wed, Dec 2, 2009 at 4:57 AM, François Legal wrote: Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale@hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [12] iax:guest@conference.freeswitch.org/888 [13] googletalk:conf+888@conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel@thom.fr.eu.org [2] mailto:FreeSWITCH-users@lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale@hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale@gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888@conference.freeswitch.org [13] http://iax:guest@conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org

1 0

call barge in
by Nikolay Kondratyev 03 Dec '09

03 Dec '09

Hi all, I'm evaluating FS for our organization. I must fulfill the following requirements: 1. Call recording: All (or selected) calls to the secretary must be recorded. 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes "emergency" (for example, an extension with emergency prefix) call to one of these subscribers -> Secretary barges in the established call (conference). 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. Can anybody please advise if this is possible with FS? If yes, is it just a configuration task, or additional programming will be needed? Thanks in advance, Nikolay.

4 4

Re: [Freeswitch-users] Remote fetching of voicemail
by François Legal 03 Dec '09

03 Dec '09

No, my voicemail extension (I have 2 actually) is called vmain_unregistered_user, so in voicemail.conf.xml I have : <param name="vmain_extension" value="vmain_unregistered_user"/> But still (and I don't even know if I'm using it the right way), I would expect that when the voicemail greeting starts, I could press * on the phone to call the vmain_unregistered_user extension in my dialplan, but that never happens. What I want, is when someone dials in (this is usefull only from outside so via FXO line) his extension (or any other, it's not important) that when he reach the voicemail to leave a message, override the "leave a message" voicemail and enter the "check my messages" voicemail (with an authentication step). I'm not sure this is clear. François On Wed, 2 Dec 2009 12:16:14 -0500, Frank Carmickle <frank(a)carmickle.com> wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> I did check (and modify as my voicemail extension name is not vmain) the >> voicemail.conf.xml, and vmain-key is *. >> >> What I mean by remote fetching of voicemail, is that someone may dial in, >> either from inside (via FXS or even SIP) or outside (via FXO), then when >> reaching the voice mail to leave a message, he could dial some specific >> digit (or digits) to reach the voicemail login and fetch his voice mails. >> >> I can do this using bind_meta_app (it is already working), but then I >> need >> to terminate the extension when invoking the meta_app, otherwise >> freeswitch >> may sometimes hang if the meta app is called after the "leave a message" >> voicemail tone. > > Alright. I missed what vmain actually does in the voicemail config. It > actually calls the extension named vmain in the dialplan. So if you don't > have this then you will need to have one. Thanks for asking this question > because my voicemail auth was broken and I didn't even know it! I fixed it > and a working extension for vmain can look like this. > > <extension name="vmain"> > <condition field="destination_number" > expression="^vmain$|^voicemail$|^\*98$|^\*86$"> > <action application="answer"/> > <action application="set" data="voicemail_authorized=false"/> > <action application="sleep" data="200"/> > <action application="voicemail" data="check default $${domain} > ${effective_caller_id_number}"/> > </condition> > </extension> > > HTH > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

3 4

How do I know the destination profile name?
by Eli Hayun 03 Dec '09

03 Dec '09

Hi We have more then one profile. To make a call I have to enter : bridge sofia/profile/number@ip The problem is when I use : "${use_profile}" I am getting the caller profile, and I need the destination profile. How do I get this information? Thanks Eli

5 9

First steps in FreeSWITCH
by Martin Rodriguez 02 Dec '09

02 Dec '09

Hi list; I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and 10 years in VoIP (Cisco). I need a reference guide to start working with FreeSWITCH. I download the official documentation, it would need some other configuration examples and dialplan sip device for calls. Martin Rodriguez

2 1

CDR records
by Puskás Zsolt 02 Dec '09

02 Dec '09

Hi Guys! I'm using the latest svn (15711) with the default xml config. Only modified cdr_csv.conf.xml the line <param name="legs" value="a"/> to <param name="legs" value="ab"/> Here is what i do: 1. 1000 calls 1001 (1001 answers the call) 2. 1001 do blind transfer to 1002 (using *1) 3. 1001 hangs up 4. 1002 answers the call 5. 1002 and 1000 hangs up 3 cdr records are generated (simplified): from,to,start,duration "1000" "1001" "2009-11-29 15:21:53" "53" "50" "1000" "1002" "2009-11-29 15:21:53" "79" "76" "1000" "1002" "2009-11-29 15:22:46" "26" "23" As you can see the second cdr is incorrect because 1000 doesn't speak with 1002 for 76 second. Is this a normal ? Is it possible to make only 2 record ? Thank you for any answer.

3 5

Transfer Problem
by Jonathan Barou 02 Dec '09

02 Dec '09

Hi everybody, I'm actually using the lastest version of Freeswitch, I have a problem. I have a trunk SIP with my PABX. There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) 2. one Alcatel IpTouch 4028 with number 987 (on PABX) 3. one Siemens Gigaset A580 IP with number 8401 (on Freeswitch) *The first test* is to say to the phone 2 to transfer all the call to number 8401. So when I dial 987 on the phone 1, all work perfectly, the phone 3 is ringing and it's work. I have that in the log : 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/368(a)10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 port 32000 codec: 8 ms: 90 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 23054 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/368(a)10.33.69.246! 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ sip:8401@10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ 368(a)10.33.69.246 skipping state [early][183] 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/368(a)10.33.69.246 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16@8000hz 1 channel 90ms 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ 368(a)10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port confirmed. 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read Buffer at 1440 bytes vs 81 *The Second Tes*t is to say to the phone 1 to transfer all the call to number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just one time and after it hangup. I have that in the log : 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/987(a)10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 port 32000 codec: 8 ms: 90 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 27732 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/987(a)10.33.69.246! 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ sip:8401@10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ 987(a)10.33.69.246 skipping state [early][183] 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/987(a)10.33.69.246 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16@8000hz 1 channel 90ms 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ 987(a)10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ 987(a)10.33.69.246 entering state [terminated][487] 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ 987(a)10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/987(a)10.33.69.246 [KILL] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987(a)10.33.69.246 [BREAK] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup sofia/internal/sip:8401@10.33.170.231:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [KILL] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401@10.33.170.231:5060) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401@10.33.170.231:5060) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ sip:8401@10.33.170.231:5060 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ sip:8401@10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/internal/sip:8401@10.33.170.231:5060 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:8401@10.33.170.231:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401@10.33.170.231:5060) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:8401@10.33.170.231:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401@10.33.170.231:5060) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401@10.33.170.231:5060) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:8401@10.33.170.231:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401@10.33.170.231:5060) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:8401@10.33.170.231:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 (sofia/internal/sip:8401@10.33.170.231:5060) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401@10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 (sofia/internal/sip:8401@10.33.170.231:5060) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/sip:8401@10.33.170.231:5060 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/sip:8401@10.33.170.231:5060) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401@10.33.170.231:5060) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ sip:8401@10.33.170.231:5060 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:8401@10.33.170.231:5060 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401@10.33.170.231:5060) State DESTROY going to sleep 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. Cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/987(a)10.33.69.246) State EXECUTE going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987(a)10.33.69.246) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987(a)10.33.69.246) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ 987(a)10.33.69.246 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ 987(a)10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/987(a)10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987(a)10.33.69.246) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/987(a)10.33.69.246) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987(a)10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987(a)10.33.69.246) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987(a)10.33.69.246) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/987(a)10.33.69.246 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987(a)10.33.69.246) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/987(a)10.33.69.246) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987(a)10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 (sofia/internal/987(a)10.33.69.246) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 (sofia/internal/987(a)10.33.69.246) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/987(a)10.33.69.246 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/987(a)10.33.69.246) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987(a)10.33.69.246) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ 987(a)10.33.69.246 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/987(a)10.33.69.246 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987(a)10.33.69.246) State DESTROY going to sleep Finally when I tried to call the phone 3 with the phone 1 it's working, and not when I want to call the phone 3 with the phone 2, like just before, it's ringing just one time and hangup. Thanks you. Best Regards -- John

2 4

Dictation System
by David Laperle 02 Dec '09

02 Dec '09

Hi Freeswitch users, i'm new into the PBX world. I just installed FreeSwitch and made work great, but one of my goal with the PBX system is to use it as a dictation system. We were using Callweaver, and there's a Dictation module for CW and one for Asterisk, but i can't find one for FreeSwitch so far. Is there anything in the trunk that i could find or any work in progress? I'm willing to develop a bit to help the work in progress, i have the programming knowledge but not the VOIP/PBX knowledge, so a work in progress could be enough for me to start with and complete the work! If any of you have an idea or a hint for me i would be very grateful! Thanks a lot, David Laperle Administrateur réseau / Network administrator (514) 393-7647 dlaperle(a)rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montréal Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil�gi� et confidentiel est destin� � la seule personne ou entit� � laquelle il est adress�. Pour toute autre personne, toute action prise en rapport � ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'�tes pas la personne autoris�e � recevoir ce courriel, S.V.P. le retourner � l'exp�diteur et le d�truire. Bien que ce courriel ait �t� trait� contre les virus, il est de la responsabilit� du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot�g�s par le secret professionnel. Si vous d�sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier �lectronique ordinaire non s�curis�, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you.

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Remote fetching of voicemail
by François Legal 02 Dec '09

02 Dec '09

Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks François

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