- FreeSWITCH-users - FreeSWITCH Mailing Lists

Need Help In IVR
by Thangappan.M 05 Aug '09

05 Aug '09

Dear all, I am in the process of implementing IVR in Perl using outbound socket. In the case of the XML macro I can easily specify the timeout,inter digit timeout value as <menu name="demo_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout ="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/> </menu> Is there any way for Perl to configure this values. Where are the variables resides? I am struggling to implement this? -- Regards, Thangappan.M

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pocketsphinx
by Helmut Kuper 05 Aug '09

05 Aug '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/) which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKVzne4tZeNddg3dwRAthzAJ4hvonJLgaTWc3kCQXhESb2wsTu8QCeP/DD skUOkNgUHLRaKqVGWWk1uM8= =HkcN -----END PGP SIGNATURE-----

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freepbx for freeswitch
by Giovanni Maruzzelli 05 Aug '09

05 Aug '09

Yay! http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future Darren Schreiber has made the announcement and is doinng a presentation of FreePBX V3 right now at www.cluecon.com.

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q: fresh installation: Couldnt open /usr/local/freeswitch/conf/freeswitch.xml
by tom 05 Aug '09

05 Aug '09

hi just installed freeswitch via svn. - bootstrap - configure - make install - ./freeswitch gives me: acerdebian:/usr/local/freeswitch/bin# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance.... 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory) Cannot Initialize [Cannot Open log directory or XML Root!] bump - help thx

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How to change the contact when fs sending REGISTER??
by Brad Tuan 05 Aug '09

05 Aug '09

As title ,I know how to do when sending INVITE but how to do it when fs sending REGISTER?? For example , when gateway registering , the contact is gw+abcd(a)XXX.XXX.XXX.XXX , how to change it to *abcd(a)XXX.XXX.XXX.XXX??* ** *Please help* ** **

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Re: [Freeswitch-users] Looking for some FreeSWITCH job
by Otto 05 Aug '09

05 Aug '09

Diego, You can contact me. Br. Otto van Haaren www.5ca.com <http://www.5ca.com/> ovh(a)5ca.com ________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Diego Viola Sent: dinsdag 4 augustus 2009 23:22 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Looking for some FreeSWITCH job Hi, I'm currently looking for some FS jobs, I really need one, I'm currently unemployed and looking for some serious FreeSWITCH jobs. Anyone? P.S: I also do FS and web development with any language, PHP, Ruby, etc. anything really. Thanks, Diego

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Best practice for inbound calls with scripting
by Greg Thoen 05 Aug '09

05 Aug '09

Hi. I am setting up a large inbound only system with multiple DIDs coming in; each call is processed fairly intensively with db lookups, wav files played, pocketsphinx is used, wav files recorded, etc. Before I get too far down one path, I was wondering if anyone had any insight into the best, most scaleable way to do this out of the several methods I can do: 1 Call comes in dialplan calls specific javascript based on DID javascript uses ODBC to pull info from local mysql db call is handled in javascript 2 Call comes in dialplan calls specific javascript based on DID javascript uses CURL to get info from local mysql db call is handled in javascript 3 Call comes in php socket is listening for call php script runs, pulling info from mysql db call is handled in php using esl.php 4 Call comes in dialplan calls specific lua script based on DID lua uses luasql.mysql to get info from local mysql db call is handled in lua using lua api 5 Call comes in xml_curl is used for dynamic dialplan js called, continues like #1 I know that they will all do essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks. -- Greg

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FreeSWITCH code to emulate Asterisk dialing announcement
by Michael Frager 05 Aug '09

05 Aug '09

Hello, I'm in the process of moving my VOIP application from Asterisk to FreeSWITCH. I was wondering if it is possible to emulate the call announcement feature that is available on Asterisk. On Asterisk it looks like this, with the "A(...)" parameter: Dial(SIP/15555551212|180|A(connecttone1)) Note that this announcement is only played for the called party, the calling party does NOT hear the tone. I'm guessing this can be done with FreeSWITCH. Does anyone know how I might accomplish this? Thanks in advance, -Mike Fragre

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Qualify IM comm. across two distinct SIP domains
by David Nembrot 05 Aug '09

05 Aug '09

Hi everybody, I've just configured two Freeswitch servers (FS#1 and FS#2) to enable SIP communications between their two distinct SIP domains. The fact is that the IP telephony is up & running across these two domains, but the IM doesn't seem to be operational... Basically the dialplan for telephony has been ticked in a satisfactory way, my guess is that I'm missing something in the config files in order to enable IM services throughout the two domains.. Since they are in different networks, it seems reasonable for example to force the IM comm. get through the FS#1 so to reach FS#2 domain... hence my question: How can I shape up a kind of "IM dialplan" on my servers ? (i.e. set up the routing aspects for IM too) or How can I get this IM stuff configured differently? Thanx & Regards, David N. Hi everybody, I've just configured two Freeswitch servers (FS#1 and FS#2) to enable SIP communications between their two distinct SIP domains. The fact is that the IP telephony is up & running across these two domains, but the IM doesn't seem to be operational... Basically the dialplan for telephony has been ticked in a satisfactory way, my guess is that I'm missing something in the config files in order to enable IM services throughout the two domains.. Since they are in different networks, it seems reasonable for example to force the IM comm. get through the FS#1 so to reach FS#2 domain... hence my question: How can I shape up a kind of "IM dialplan" on my servers ? (i.e. set up the routing aspects for IM too) or How can I get this IM stuff configured differently? Thanx & Regards, David N. Hi everybody, I've just configured two Freeswitch servers (FS#1 and FS#2) to enable SIP communications between their two distinct SIP domains. The fact is that the IP telephony is up & running across these two domains, but the IM doesn't seem to be operational... Basically the dialplan for telephony has been ticked in a satisfactory way, my guess is that I'm missing something in the config files in order to enable IM services throughout the two domains.. Since they are in different networks, it seems reasonable for example to force the IM comm. get through the FS#1 so to reach FS#2 domain... hence my question: How can I shape up a kind of "IM dialplan" on my servers ? (i.e. set up the routing aspects for IM too) or How can I get this IM stuff configured differently? Thanx & Regards, David N. [PiÃ¨ce jointe retirÃ©eÂ : type d'origine de la piÃ¨ce jointe: "text/html", nom: "Version_HTML_du_message"]

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Qualify IM comm. across two distinct SIP domains
by David Nembrot 05 Aug '09

05 Aug '09

Hi everybody, I've just configured two Freeswitch servers (FS#1 and FS#2) to enable SIP communications between their two distinct SIP domains. The fact is that the IP telephony is up & running across these two domains, but the IM doesn't seem to be operational... Basically the dialplan for telephony has been ticked in a satisfactory way, my guess is that I'm missing something in the config files in order to enable IM services throughout the two domains.. Since they are in different networks, it seems reasonable for example to force the IM comm. get through the FS#1 so to reach FS#2 domain... hence my question: How can I shape up a kind of "IM dialplan" on my servers ? (i.e. set up the routing aspects for IM too) or How can I get this IM stuff configured differently? Thanx & Regards, David N. Hi everybody, I've just configured two Freeswitch servers (FS#1 and FS#2) to enable SIP communications between their two distinct SIP domains. The fact is that the IP telephony is up & running across these two domains, but the IM doesn't seem to be operational... Basically the dialplan for telephony has been ticked in a satisfactory way, my guess is that I'm missing something in the config files in order to enable IM services throughout the two domains.. Since they are in different networks, it seems reasonable for example to force the IM comm. get through the FS#1 so to reach FS#2 domain... hence my question: How can I shape up a kind of "IM dialplan" on my servers ? (i.e. set up the routing aspects for IM too) or How can I get this IM stuff configured differently? Thanx & Regards, David N.

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