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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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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

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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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Posibility to build a module for chan dongle
by Josue Diaz Cruz 03 Jan '12

03 Jan '12

I was thinking if it could be possible to build a module like the asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place calls throught GSM networks I can build an asterisk server for this, but i am working continously with freeswitch and i have not too much knowledge of c/c++. I enjoy really work with freeswitch. I think that make this module could be nice to use it in a low cost sip to GSM gateway. We can use it to develop pbx systems with gsm, or trunking. Perhaps there is other module better than this made in freeswitch and i am asking an stupid thing. What i like from this chan dongle is that it is not needed to use any sound card or other device for this. Just only as a module cause the huawey have the voice mode activable. The web page for the chan is this: http://wiki.e1550.mobi/doku.php?id=installation It is open source. so we can use the sources without any issue. Sorry me if the information is not really exaustive or if i maid many mistakes not knowing completely the posibilities with other modules that can do this job. By the way, sorry me for my english. It is not really good. Josue Diaz Cruz Departamento Tecnico y Soporte <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia <http://www.coinfru.com/> www.coinfru.com

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vBilling Announcement - First Beta Release
by Muhammad Naseer Bhatti 03 Jan '12

03 Jan '12

Hello everyone, Though a few days late, but happy new year to everyone. Last year was tough and we worked hard to get vBilling up and running. So, here we are, announcing the first beta release. vBilling is the first Open Source billing platform for FreeSWITCH. Take a look at http://www.vbilling.org/ for more details. Some of the major features includes: Both Pre-Paid and Post-Paid model Multiple administration access levels Multiple reseller level Easy rate/price management Route management Separate user interface to view their CDR(s) and billing information Authentication by IP/ANI and SIP registration Codec management for both user and switch CDR statistics Gateway statistics Admin/Reseller/User management Switch management from 1 GUI Balance and payment information I would like to invite everyone to download and install using the installer scripts available at http://vbilling.org/get-started/ We need your contribution and help to make this a success project. Don't forget to visit our forum at http://forum.vbilling.org/ and also the Customer support http://support.vbilling.org/ if required. All the source code is also available http://github.com/digitallinx/vBilling/ So, take a look, download, install and have fun. Thanks, Muhammad Naseer (Goni)

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