May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch 1.0.8 likely release date?
by Sidharth Kshatriya 12 May '11

12 May '11

On the freeswitch users list way back in Jan 2011, it was mentioned that 1.0.8 can be expected in a month or so. Its May now :-) Can anyone tell us approximately when 1.0.8 is likely to be released... even a ball park? Thanks, Sidharth -- Sidharth Kshatriya www.sidk.info

2 1

FreeSwitch video is impossible to do MCU and H.239 in H.323, with SIP still there is no BFCP
by Shamun 12 May '11

12 May '11

Hello Team, FreeSWITCH does not work for Video. To do Video with FreeSWITCH, Its simply a great nightmare. None of our equipment is compatible with FreeSWITCH we have tried mod_h323/mod_opal and regular SIP from FreeSWITCH (none of them works with h.323). - Is it possible to do FreeSWITCH IVVR (Flash or MPEG4 or JPEG) ? - Will FreeSWITCH support H.323 (and its related family such as H.460, H.239, etc etc), there are lot of staff just does gets compatible with SIP (specially our hardwares, i believe same with others). - Will FreeSWITCH allow following equipment supports + more other equipments who really depends on mod_h323/mod_opal (SIP has less hope with hardware's compatibility) Tandberg Edge 85 MXP Tandberg C90 VCON vPoint HD 10.0 Radvision Thanks & Regards Shamun

2 1

execute_on_answer_n and execute_on_answer
by Yehavi Bourvine 12 May '11

12 May '11

Hello, According to the wiki, starting from April there is an additional syntax for execute_on_answer which is execute_on_answer_1 (and 2 and on). If I want to use the new syntax, how shall I do it? Use only execute_on_answer_1,2,... or first use execute_on_answer and the following statements will use the numbered format? What happens if I mix the two? What happens if I miss a number (i.e. use _1 and then _3)? Thanks! __Yehavi:

2 1

mod_say in different codecs
by David Notivol 12 May '11

12 May '11

Hi all, I'm trying to setup an IVR server, and I'm using the session:say function from a LUA script. My question is if there's any way of having mod_say playing audios different than audio files; I mean audios encoded in G729, G711ulaw, G711alaw, etc. to avoid having FS making transcoding every time I run a say command. Checking the folders tree for the sounds, I can see the place for the different languages, voices, and wav qualities (8k, 16k...) is clearly specified; but is it a way to place files encoded in different codecs? Thanks in advance. Regards, David Notivol

3 4

Primary leg hangs up when other leg rejects the call
by lakshmanan ganapathy 12 May '11

12 May '11

Dear all, I'm using Event Inbound Socket. I want to originate a call to the user and bridge him with different user. So I used the following commands from the telnet. api originate {exec_after_bridge_app=park,ignore_early_media=true}user/1010 &park() Content-Type: api/response Content-Length: 41 +OK cf9a78a2-7aff-11e0-b732-4f9b0298a6ed sendmsg cf9a78a2-7aff-11e0-b732-4f9b0298a6ed call-command: execute execute-app-name: bridge execute-app-arg:user/1005 Content-Type: command/reply Reply-Text: +OK The user 1005 rejects the call and I got the response as CALL_REJECTED. I expect that primary leg will not hangup and go into park mode. But it got hangup. Since by default hangup_after_bridge will be false, I'm not setting it explicitly here. FreeSwitch logs are here, http://pastebin.freeswitch.org/16264 Can someone please tell me what is the problem I've done and how to make the primary leg again in park() ??

2 3

Receiving calls from an external IVR
by Ira Tessler 11 May '11

11 May '11

I would like to integrate Freeswitch into my existing IVR platform. For now, I would like to use it to register VoIP phones, handle outbound calls and receive inbound calls from my IVR. We are a service provider, so this would be a multi-tenant configuration. Each one of our customers would be set up in their own Freeswitch context. I would set up the same extensions for a customer in our IVR and for Freeswitch. When someone dials ext 101 on our IVR, I would like the IVR to make a call to Freeswitch and have ext 101 ring for the customer’s context. My IVR would pass in the callerid of the inbound caller and I would like that number to display on the VoIP phone as the callerid and name. Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to make a call to the IVR. Is this possible? Are they any examples out there? How can I get started? I am a newbie to Freeswitch. Thanks, Ira Tessler

3 4

database authentication
by Michel Freiha 11 May '11

11 May '11

Dear All, I have a question that I'm not be able to find an answer or I'm a bit confused about listed details regarding this topic...As you know, OpenSips or asterisk has real time authentication which mean all endpoints credentials are defind into a database and when a user with a specific extension tries to register to registrar server, this server send an authentication to database in order to accept registration or not.. Is this scenario doable with freeswitch or not? Regards

2 1

What debugs to enable in order to trace tone_detect problems
by Yehavi Bourvine 11 May '11

11 May '11

Hello, I am trying a new AudioCodes mediant-1000 PRI gateway (instead of a cisco gateway). I have a problem with tone_detect: I enable it with execute_on_answer and it works only if the call gets to voicemail. If I answer I hear the first beep and that's all. The call is left connected, but it does not activate the application set by tone_detect. what debugs shall I enable in order to see what tone_detect is doing? Thanks! __Yehavi:

1 0

FreeSWITCH Conference Call Today
by Michael Collins 11 May '11

11 May '11

Hello gang! We have a light agenda again today, however I still want everyone to call in so that we can discuss a few topics. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_11 Be sure to add any items that you want to discuss today. -Michael

1 0

mod_lua play file into conference
by Josh M. Patten 11 May '11

11 May '11

OK so I've successfully managed to get a lua script working to enhance the functionality of the "mad boss" intercom via conference example however I'm having trouble getting an audio file to stream into the conference at the beginning. Here is what I'm trying: Dialplan: <extension name="outbound-socket"> <condition field="destination_number" expression="^3402$"> <action application="lua" data="/usr/local/freeswitch/scripts/test.lua 3402"/> </condition> </extension> Snip of test.lua: session:execute("conference", argv[1] .. "@default") session:execute("set", "tmp=${conference " .. argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav}") I've also tried: session:execute("conference", argv[1] .. "@default") api = freeswitch.API() confplay = api:execute("conference", argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav ") In watching fs_cli it appears freeswitch never executes the entry that would set this file. What am I doing wrong here? I've got the FreeSWITCH 1.0.6 book and I can't seem to find my answer there either. Oh, playing a sound on participant entry won't work because it will play that sound for EVERY user that enters the conference. When 80 extensions are auto-joined to the conference at the same time it is useless for a while playing entry sounds for all those extensions. Thanks! Josh Patten Brazos County Network Engineer 979.361.4676

4 12