January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

error on git update
by Charles Bujold 04 Jan '12

04 Jan '12

Tried updating Freeswitch (last update from GIT done Dec 1 2010) today and I'm getting the following error. Can somebody identify what needs to be done to get a proper update. Using Ubuntu 10.4 server. Never had a problem before! In file included from /usr/src/freeswitch.git/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch.git/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.git' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.git' make: *** [current] Error 2 Thanks CJB

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timeout for curl
by Yuriy Nasida 04 Jan '12

04 Jan '12

Hello FS users, I do some curl request from my dialplan. It looks like this:<action application="set" data="lookup=${curl(http://mycnameprovider.com/cname.php?number=${myvar})}" inline="true"/>If cname provider doesn't answer for long time, I have issue because FS doesn't process a call.whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, FS must continue the logic of dialplan. I read about mod_cidlookup but it looks like that this module have same issuehttp://wiki.freeswitch.org/wiki/Mod_cidlookup Please advise Thanks.

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Voicemail - see if user left a message?
by Avi Marcus 03 Jan '12

03 Jan '12

Can the xml_cdrs tell me if the user actually left a voicemail, rather than merely that VM picked up? I don't see where. I know it's on ESL, and it hits the voicemail email code.. what's the easiest way to integrate it with the CDRs? preferably with the xml_cdrs? Maybe a patch to the xml_cdr mod? Or the easiest is to have mod_voicemail set a channel variable once a voicemail gets marked as saved or the like? Is there currently a solution for this that I missed? Thanks, -Avi

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conf count when entering room
by Rodney 03 Jan '12

03 Jan '12

MC, I have not figured out how to get the conference count upon entering the room. I would imaging loading the Conference Announce Count extension at the same time as dropping into the conference. I am unsure of how to do this so it gets passed the conference number i am entering. I would imagine this would be a good start: ? <extension name="room1"> <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/>  <action application="clear_digit_action" data="all"/> <action application="bind_digit_action" data="moderator,5,exec:transfer,555 XML default"/> <action application="bind_digit_action" data="moderator,6,exec:transfer,319 XML default"/> <action application="bind_digit_action" data="moderator,7,exec:transfer,302 XML default"/> <action application="bind_digit_action" data="moderator,#,exec:transfer,759 XML default"/> <action application="bind_digit_action" data="moderator,0,exec:transfer,300 XML default"/> <action application="digit_action_set_realm" data="moderator"/> <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav"/>  <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition></extension> ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 3, 2012 4:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 20 ----- 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: conf count (Michael Collins) 2. Re: Radio stream binding ? (Michael Collins) 3. Re: Radio stream binding ? (Rob Hutton) 4. Re: int/ext dial tones not synchronous (Michael Collins) 5. timeout for curl (Yuriy Nasida) 6. Re: Silence ESL event (Michael Collins) Did you figure this one out yet? -MC On Sun, Dec 25, 2011 at 8:20 AM, Rodney <notlikeme75(a)yahoo.com> wrote: > >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. > >_________________________________________________________________________ >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 > > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: but the station i need is this > >mmsh://live.sis.gov.eg/live?MSWMExt=.asf > >and it is not there. I would appreciate your help > Bummer. ASF is a proprietary Microsoft technology, so I would contact the people using it and let them know that they are limiting their potential audience by using a locked-down container format. If you *really* want this then you can probably hire someone to do a workaround. My guess is that you'll need to pay a fair amount because the people with the requisite skills to do this will be in high demand. -MC If you can get permission to rebroadcast, you could set up a machine to transcode it into a format that is supportable... On Tuesday 03 January 2012 15:44:34 Michael Collins wrote: > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran <sherifomran2000(a)yahoo.com>wrote: > > > but the station i need is this > > > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > > > and it is not there. I would appreciate your help > > > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > I don't believe you can reliable put these tones in synch. I've never heard of any telephony system that does so, although I admit my experience is limited to legacy PBXes and FreeSWITCH. -MC On Fri, Dec 23, 2011 at 8:47 PM, <georg(a)riseup.net> wrote: Hi, > >> I'm not sure that I understand what the issue is here. What two "tones" >> are >> not in sync? You said "dial tone" but I'm pretty sure that's not what you >> mean. Are you talking about the ringback tone that the caller hears? > >Exactly Michael, I was talking about the ringback tone. >I didn't know the correct expression and used a translation engine... > >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 > Hello FS users, I do some curl request from my dialplan. It looks like this: <action application="set" data="lookup=${curl(http://mycnameprovider.com/cname.php?number=${myvar})}" inline="true"/> If cname provider doesn't answer for long time, I have issue because FS doesn't process a call. whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, FS must continue the logic of dialplan. I read about mod_cidlookup but it looks like that this module have same issue http://wiki.freeswitch.org/wiki/Mod_cidlookup Please advise Thanks. You might be able to produce the same behavior with the "wait_for_silence" dp application. -MC On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar <hynek.cihlar(a)gmail.com> wrote: 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 > >_________________________________________________________________________ >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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codec-ms for specific gateway
by Anestis Mavro 03 Jan '12

03 Jan '12

Hi, I need to set a different packetization time for one specific gateway, keeping the default settings for other gateways. I have tried the parameter codec-ms (which is for sofia.conf.xml) in the settings for the specific gateway, but it doesn't work. Is there any other way to achieve this? Thank you Anestis

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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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freeswitch as MGW
by Juan Pablo L 03 Jan '12

03 Jan '12

Hello all, i was wondering if its possible the make a configuration in such a way that you have one box, box1, with a freeswitch instance to deal with all the sip related tasks and a next box, box2, that deals with all the audio related tasks being controlled by box1 by means of MGCP, or any other protocol for that purpose, acting as a MGW (that could also be used by other nodes in the network) ?

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freeswitch b-leg data logging
by Sherif Omran 03 Jan '12

03 Jan '12

Hi I have a problem with freeswitch and don't get the b-leg information although it is enabled in autoload_configs/xml_cdr.conf.xml file and would appreciate if you could help. waiting your reply kind regards, Sherif Omran

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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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