December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

transfer to dynamic con from static
by Rodney 04 Jan '12

04 Jan '12

everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? <control action="execute_application" digits="#" data="transfer 759 xml default"/> `rodney ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, December 22, 2011 6:30 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 145 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Some questions regarding IPs and gateway (Michael Collins) 2. transfer conference to conference (Rodney) 3. Re: Odp: transfer to conference from conference / caller controls (Anthony Minessale) The "domain" method you used is actually a good way to go if you're always wanting the same IP for all network comms. -MC On Thu, Dec 22, 2011 at 3:19 PM, <georg(a)riseup.net> wrote: > You want to set the IP address in the SIP profile. Remember, FreeSWITCH >> can >> use multiple IP addresses and multiple ports, so it's not like Asterisk at >> all in that respect. You control which IP/Port each SIP profile uses. If >> you tell the SIP profile explicitly to use a specific IP address then >> that's what it will use and will ignore the other NICs. > >I achieved this now trough another way, see my other mail for this. >I this "a no-go"? > >I already set the ips in my internal and my external profile, however, FS >was still binding itself to the wan-ip (which is the way it normally >works, if I'm right?). My plan was to change this and exclude especially >the wan-ip. > >Georg > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( condition destination_number ^760$ 1 action answer 3 action set conference_user_list=| 10 action play_and_get_digits 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 action phrase spell,${target_num} 30 action sched_hangup +10 normal_clearing bleg 35 action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf You should be using transfer not execute_extension or you are going into the new conference mid digit parse and blocking in the first conference waiting to return. If you use the transfer app then you will exit the current conference and transfer to the new one. On Dec 22, 2011 2:42 PM, "Rodney" <notlikeme75(a)yahoo.com> wrote: Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back. thanks for your help. ~rodney > > condition destination_number ^760$ 1 > action answer 3 > action set conference_user_list=| 10 > action play_and_get_digits 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 > action phrase spell,${target_num} 30 > action sched_hangup +10 normal_clearing bleg 35 > action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf > > > > 4. Odp: transfer to conference from conference / caller controls > (Mariusz Czulada) > > >> > >Perhaps this will be useful for you: > >Dialplan: > > <extension name="bind_and_join"> > ... > <action application="bind_meta_app" data="4 a s execute_extension::sub_conf"/> > ... > <action application="conference" data="mainconf@cx"/> > ... > </extension> > > <extension name="sub_conf"> > ... > <action application="bridge" data="loopback/app=conference:${conference_name}_sub@cx_sub"/> > ... > </extension> > >conference.conf.xml: > > <group name="cx_controls"> > ... > <control action="hangup" digits="9"/> > </group> > > <group name="cx_sub_controls"> > ... > <control action="hangup" digits="4"/> > </group> > > <profile name="cx"> > ... > <param name="caller-controls" value="cx_controls"/> > </profile> > <profile name="cx_sub"> > ... > <param name="caller-controls" value="cx_sub_controls"/> > </profile> > > >Dnia 20-12-2011 o godz. 18:21 Rodney napisał(a): >I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501@default to <digits>@dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney >> >> >> <group name="default"> >> <control action="mute" digits="1"/> >> <control action="hangup" digits="0"/> >> <control action="execute_application" digits="4" data="playback C:/Program Files/FreeSWITCH/recordings/conf_help.wav"/> >> <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> >> <control action="execute_application" digits="#" data="execute_extension 759 xml default"/> >> >>- <group name="dynamic"> >> <control action="mute" digits="1"/> >> <control action="execute_application" digits="0" data="execute_extension 401 XML default"/> (main ivr menu extension) >> <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> >> >> >> >> >> >>condition destination_number ^759$ 1 >> action answer 2 >> action set conference_user_list=| 11 >> action play_and_get_digits 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 >> action phrase spell,${target_num} 16 >> action conference ${target_num}-127.0.0.1@default >> > > > >_______________________________________________ >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 > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > _______________________________________________ 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

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Silence ESL event
by Hynek Cihlar 04 Jan '12

04 Jan '12

Hi all, is there any way to receive a silence-detected event through ESL? The use case is to hangup the call in case nothing interesting is going on on the channel. Sent from my mobile device

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how do I play music on hold for A-leg while IVRing with B-leg
by Greg Buzzard 03 Jan '12

03 Jan '12

Hi All, new to FreeSwitch, great software! Read earlier posts, bought FS book and am about half-way through it and still have a question. Problem context: I'm trying to do a form of call screening for home use using a Lua script. I record a "self-intro" message from calling party (A-leg), would then like to play hold music for A-leg while creating new session and doing IVR dance with called party (B-leg). IVR objective is to see if B-leg wishes to accept, add caller-ID to whitelist or blacklist, and/or send to voicemail, etc. I have all of this working except the ability to play hold music to A-leg before either bridging them, sending them to voicemail or playing a message and dropping them. Specific problem: I don't have my head wrapped around the "clean" approach to addressing this problem. I.e., seems like I may want to "park" the A-leg (with music), while I do the IVR with B-leg. Then either: (1) unpark A-leg and bridge to B-leg or (2) hang-up B-leg, unpark A-leg and send to VM or (3) other embellishments where I add caller-ID (if it exists) to either white or black lists and/or play recording to telemarketers to leave me alone (with no VM), etc. If this is a good approach, getting a few pointers on key commands/apps to use would be great. FWIW, my first simple-minded approach to transfer caller (A-leg) to hold-music extension while connecting/IVRing with B-leg didn't work as execution of the transfer was held until the IVRing was done). Similar to what I saw in some earlier postings. If I get this figured out, I'm happy to summarize and post useful snippets to the email archive, wiki page or wherever, when done. -g

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Connecting to Event Socket
by Arnuld Uttre (Phonologies) 03 Jan '12

03 Jan '12

I mostly connect to event-socket using telnet and then subscribe to receive events. I want to connect to event-socket using a simple TCP client from within a C Program. Is it a goo idea or there is a better way ? -- Arnuld Uttre Systems Software Engineer arnuld(a)Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992

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Can not receive Events
by Arnuld Uttre (Phonologies) 03 Jan '12

03 Jan '12

I am running freeswitch. I have connected to it suing telnet on event socket. Through the same machine, using a different x-terminal, I have subscribed to: event plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain I am able to make calls using "api originate" command and got the call fine on my phone. I got +OK uuid-here too but I did not receive any events. I event tried event plain CHANNEL_DESTROY from different terminal but still no luck. I am usng freeswitch 1.0.7 on CentOS 5.5 with my own application configured in autoload_configs/myapp.conf.xml >From where I should start solving the problem ? Here are the contents of event_socket.conf.xml : <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="192.168.0.229"/> <param name="listen-port" value="8021"/> <param name="password" value="password-here"/>  </settings> </configuration> -- Arnuld Uttre Systems Software Engineer arnuld(a)Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992

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Continue after lua execute bridge
by David Villasmil 03 Jan '12

03 Jan '12

Hello Gang, I'm executing from a lua script: session:execute("bridge","sofia/gateway/" .. gw .. "/" .. out_number .. "") But I want my script to END if the bridge is succesfull, i.e.: Ringing, Answer, Busy... I've been testing, but it doesn't seem like the script is aware of the result of the bridge until AFTER it has been release by the "bridge" application... Is there ANY way to end my script in any of the scenarious mentioned before? (Ringing, Answer, Busy) Thanks a lot oh, and MERRY CHRISTMAS or Happy Holidays, or whatever suits you! ;) David

2 1

conf count
by Rodney 03 Jan '12

03 Jan '12

I can select the conf-alone sound in caller controls and even set up a bind digit for caller count once i am already in the conference using the Conference Announce Count Inline extension. but what i cant figure out is how to speak the caller count when entering the conference room. can anyone help with this? currently with conf-alone it only speaks the file when i am the only one left or there isnt anyone else in the room when i get there. i would like it to tell me the total callers of the room i am heading into. thanks.

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problem with date-time
by Lenin-Ruslan 02 Jan '12

02 Jan '12

Please help Windows 2008R2 FreeSWITCH Version 1.0.head (git-a58742d 2011-12-16 09-16-37 -0600) condition does not work <condition date-time="2011-12-29 17:00:01~2012-01-02 11:00:01"> <action application="playback" data="info.wav"/> </condition> /log 7 2011-12-30 12:06:29.570395 [DEBUG] switch_xml.c:2870 XML DateTime Check: date time[2011-12-30 12:06:29] =~ 2011-12-29 17:00:01~2012-01-02 11:00:01 (FAIL)

2 1

sangoma A200 with mod_freetdm on windows?!
by Tamer Higazi 01 Jan '12

01 Jan '12

Hi people! I got my A200 board running with 1 FXS module on Linux along with mod_freetdm, but I am facing problems getting it to run on Windows. From Sangoma I followed the instructions to set up the board on Windows7 winpipe module, which works so far. How do I get freeswitch with mod_freetdm to run on Windows that I can make use of the board (pbx) on a win machine? for any ideas, I would thank you. Tamer

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Freeswitch integrated with Openfire
by fieldpeak 30 Dec '11

30 Dec '11

Dear friends, Could anyone advise how to integrate FS work with Openfire like Asterisk can do, thanks. -- Regards, Charles

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