[Freeswitch-users] Gateway Ping

Hi, I activated the ping option on one of my gateway and I receive that kind of message : nta: sent OPTIONS (110269468) to udp/91.121.129.17:5060 nta: received 501 Not Implemented for OPTIONS (110269468) Does that mean the ping is working ? :) Regards, -- Laurent FABRE Directeur général 10,

Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed

Updated to latest and no more crash. Thanks a lot. -- Laurent FABRE Directeur général 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fa...@kirranet.com -Message d'origine- De : freeswitch-users-boun...@lists.freeswitch.o

Re: [Freeswitch-users] api_hangup_hook not actually executing the command

Thank you. Scott Brian West wrote: I would use mod_limit and not futz with global anything. /b On Jan 22, 2009, at 2:07 PM, Mathieu Rene wrote: Its global_setvar not set_global Math ___ Freeswitch-users mailing list Free

Re: [Freeswitch-users] Wideband/Ultrawideband Conference

Also if you're calling with PortAudio update ASAP... /b On Jan 23, 2009, at 6:46 PM, Michael Collins wrote: > Also, we're more likely to answer your questions if you're not running > 8kHz audio! ;) > -MC ___ Freeswitch-users mailing list Freeswitch-u

Re: [Freeswitch-users] bgapi uuid_kill question

Seems like this works running FS under Linux, but didn't for Windows version. Let me double check that on Monday. Thanks a lot. Anthony Minessale wrote: > Unique-ID > > On Fri, Jan 23, 2009 at 5:56 PM, pauld > wrote: > > Oh, > But where do I get it from? It's n

Re: [Freeswitch-users] auto dialing question ...

its the same idea where i gave you that example to call 9998 call 1234 where your extension is delivered via curl when 1234 is requested use the "read" and "transfer" apps in the xml you return, you can get it working statically first in your regular dialplan. On Fri, Jan 23, 2009 at 6:20 PM, She

Re: [Freeswitch-users] Wideband/Ultrawideband Conference

On Fri, Jan 23, 2009 at 4:37 PM, Brian West wrote: > FreeSWITCHers, >We have set the 8...@conference.freeswitch.org conference to be 32k, > This weekend I'm going to move it to be 48k by default. So dust off > your Logitech 350 headsets ( or go buy one from your favorite > retailer ). > >

[Freeswitch-users] Wideband/Ultrawideband Conference

FreeSWITCHers, We have set the 8...@conference.freeswitch.org conference to be 32k, This weekend I'm going to move it to be 48k by default. So dust off your Logitech 350 headsets ( or go buy one from your favorite retailer ). The 350 has the 16kHz pickup and the 20kHz output requir

Re: [Freeswitch-users] auto dialing question ...

Thanks Michael. Here are some really simple examples (we limit what people do through the web ... nothing really fancy ... just some good old fashion robo calls). ultimately today it looks like this extensions_table (context, exten, priority, app, appdata): campaign,100,1,ANSWER() --> answers call

[Freeswitch-users] Pylons example on the wiki

Hello, I just want to say that I have uploaded a Pylons wiki page with examples in svn. Pylons is a python web framework, which I think can help people to get started easily with xml_curl. If there is demand to extend the examples I am happy to do at. Suggestions are always welcome. You can nor

Re: [Freeswitch-users] bgapi uuid_kill question

Unique-ID On Fri, Jan 23, 2009 at 5:56 PM, pauld wrote: > Oh, > But where do I get it from? It's not seem to be available from > CHANNEL_EXECUTE event or any other channel related event. > > > Brian West wrote: > > Don't use the core_uuid.. use the session uuid. > > > > /b > > > > On Jan 23, 200

Re: [Freeswitch-users] bgapi uuid_kill question

Oh, But where do I get it from? It's not seem to be available from CHANNEL_EXECUTE event or any other channel related event. Brian West wrote: > Don't use the core_uuid.. use the session uuid. > > /b > > On Jan 23, 2009, at 5:35 PM, pauld wrote: > > >> Hi, >> Trying to kill a channel (hangup)

Re: [Freeswitch-users] auto dialing question ...

Try it from the FS CLI first to see how to do it. originate so type this in the console replacing the sofia url with one of your choice: originate sofia/default/1...@dom.com 9998 XML default when you answer the call you should hear the tetris song. so you can use any e...@context+dialplan

Re: [Freeswitch-users] auto dialing question ...

I see your dilemma. To keep things dynamic you definitely want to use your XML_CURL stuff. Like you said, nothing static. Can you post a sample call-flow in plain English? I'm curious about something. I don't want to say anything else. Just post a simple call flow: Initiate call Wait for answer/bu

Re: [Freeswitch-users] bgapi uuid_kill question

Don't use the core_uuid.. use the session uuid. /b On Jan 23, 2009, at 5:35 PM, pauld wrote: > Hi, > Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state > as > per mod_socket_event event information I use bgapi uuid_kill > > - but get response "ERR no such channel!". Same h

Re: [Freeswitch-users] auto dialing question ...

Anthony / Michael, Thanks for the quick responses. What I don't want to do is "drive the call" (by that listen on a socket ... do this on this event ... or anything else that my very limited FS foo would break) ... Just want to start it and then give it instructions on where to go. So I guess a b

[Freeswitch-users] bgapi uuid_kill question

Hi, Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state as per mod_socket_event event information I use bgapi uuid_kill - but get response "ERR no such channel!". Same happens if I use unique_id or channel_name as an argument to uuid_kill. What I am doing wrong? Help would much

Re: [Freeswitch-users] Freeswitch and an SPA941

Mark, Welcome to the FreeSWITCH Community. Check this out http://voxilla.com/tools/device-configuration-wizard/linksys-spa941-configuration-wizard Also when you post to the list please click new message, input the freeswitch-users@lists.freeswitch.org and do a new subject and click

[Freeswitch-users] Freeswitch and an SPA941

Hello - I am brand-new to VOIP and have set up freeswitch on our pFsense firewall. It works great with X-lite. I would like to hook up a linksys SPA941 to the system, and am having trouble finding any documentation on how to configure it. Where do I start? Thanks - Mark

Re: [Freeswitch-users] auto dialing question ...

Does AST mean Asterisk Open Source PBX ? If so, then yes I am familiar with it's archetechure as I am a former developer from that project. You have 3 choices with FreeSWITCH 1) You can open a dedicated connection to mod_event_socket or XMLRPC per call and issue the originate command from there:

Re: [Freeswitch-users] My fs_cli stopped working, can you help me please?

just for kicks, can you try a raw telnet session, just to make sure the the event socket is working properly? telnet localhost 8021 auth ClueCon (press enter twice) If you get something like this: h-3.2# telnet localhost 8021 Trying ::1... telnet: connect to address ::1: Connection refused Trying

Re: [Freeswitch-users] Auto dialing ...

Does AST mean Asterisk Open Source PBX ? If so, then yes I am familiar with it's archetechure as I am a former developer from that project. You have 3 choices with FreeSWITCH 1) You can open a dedicated connection to mod_event_socket or XMLRPC per call and issue the originate command from there:

Re: [Freeswitch-users] Auto dialing ...

http://wiki.freeswitch.org/wiki/Mod_commands#originate /b On Jan 23, 2009, at 4:33 PM, Arnaldo de Moraes Pereira wrote: On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey wrote: OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.f

Re: [Freeswitch-users] auto dialing question ...

On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey wrote: > Sorry for the double post ... actually hit send too early ... > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and others > ... nothing useful. >

Re: [Freeswitch-users] Auto dialing ...

On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey wrote: > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and > others ... nothing useful. > Today I have a platform for auto dialing with AST (centrally m

[Freeswitch-users] auto dialing question ...

Sorry for the double post ... actually hit send too early ... OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful. Today I have a platform for auto dialing with AST (centrally ma

[Freeswitch-users] My fs_cli stopped working, can you help me please?

I use the stable version of fs 1.0.2 testing out different options of configuration; Last thing I was doing was trying to fix some issues with dingaling-google talk cause i hear no audio at all from an external ip, and i still didnt get it to work; so i tried deleting the sofia_* files on the fold

[Freeswitch-users] Auto dialing ...

OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful. Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this: -- Remote machin

Re: [Freeswitch-users] Gateway Params

now all of the above is true in latest trunk On Fri, Jan 23, 2009 at 10:32 AM, Brian West wrote: > Actually now that I think of it.. we might do both in gateways now. Let me > dig into the code. > /b > > On Jan 23, 2009, at 10:39 AM, Laurent Fabre wrote: > > Thanks. That's what I was afraid of.

Re: [Freeswitch-users] mod_g729

On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles wrote: > Hi Dave, > > Down here in Brazil, the bandwidth costs is very high (around U$ 400.00/Mb) > so it should be valid only for a "non" third > world country. > G729 and G723.1 is almost a law here, if you don't play at least with G729 > you

Re: [Freeswitch-users] mod_g729

Hi Dave, Down here in Brazil, the bandwidth costs is very high (around U$ 400.00/Mb) so it should be valid only for a "non" third world country. G729 and G723.1 is almost a law here, if you don't play at least with G729 your ITSP is out of mark share! My 2 cents from a third world country. Re

Re: [Freeswitch-users] Few question regarding move from Asterisk to FS - resend

FYI, sorry, I responded to the other email first! Per Mathieu's replies, yes FreeSWITCH can do all of those things that you mentioned. The key for you will be to unlearn "the Asterisk way" because much of the way Asterisk does things is a result of working around inherent limitations in the system

Re: [Freeswitch-users] Few question regarding move from Asterisk to FS

On Fri, Jan 23, 2009 at 3:53 AM, Ivica Samija wrote: > Hi all, > our company have implemented two Asterisk servers to: > - connect two company sites > - transition to IP telephony > - cut down TCO regarding telephony > > Our interconnection schema: > > --T1/E1 provider1--< > > --T1/E1 pr

Re: [Freeswitch-users] record session in fifo

please test latest trunk. Patch added to pause media bugs while not in a bridge which should pause recordings and cut out the moh. On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke wrote: > Hello, > > we would like to distribute calls with fifo and record these sessions > but we'd like to skip the r

Re: [Freeswitch-users] record session in fifo

Tamas, These are very specific requirements. FreeSWITCH certainly has the tools necessary to do it all but it requires some knowledge and skill on your part. I think you are right to be looking at the event socket for this. You need some sort of 3PCC - 3rd Party Call Control - which is most likely

Re: [Freeswitch-users] Few question regarding move from Asterisk to FS - resend

1) A lot of people use openzap in production environements 2) probably, even if openzap doesnt implement it (which I think it does), you can use call groups to achieve the same results 3) freeswitch has a db application/api that does the same. 4) it sets the "nibble_total_billed" channel variable,

Re: [Freeswitch-users] Conference and socket outbound

On Jan 23, 2009, at 11:04 AM, Dennis wrote: > is it possible to define a profile and its params for a conference > dynamically over socket outbound? > > in the moment, if we want to have multiple profiles for different > clients, we (have to) setup a profile in the conference.conf - > otherwise w

[Freeswitch-users] Few question regarding move from Asterisk to FS - resend

Last message was incomplete, sorry for that, resending. Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Aste

[Freeswitch-users] Few question regarding move from Asterisk to FS

Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider

[Freeswitch-users] record session in fifo

Hello, we would like to distribute calls with fifo and record these sessions but we'd like to skip the recording while the caller is waiting. (we don't need to record the hold music, just the speech with the fifo consumer.) I tried but it doesn't work because the channel is answered immediat

Re: [Freeswitch-users] Gateway Params

Actually now that I think of it.. we might do both in gateways now. Let me dig into the code. /b On Jan 23, 2009, at 10:39 AM, Laurent Fabre wrote: Thanks. That’s what I was afraid of. Can I add variables inside the gateway or it’s not supported either ? _

Re: [Freeswitch-users] Gateway Params

Thanks. That's what I was afraid of. Can I add variables inside the gateway or it's not supported either ? -- Laurent FABRE Directeur général 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fa...@kirranet.com De : f

Re: [Freeswitch-users] Gateway Params

If its not done exactly like it says in the defaults ie name="asterlink.com">gateway> it WILL not parse correctly. I think we might wanna change this to be inside a params tag... but we have to be backwards compatible too. /b On Jan 23, 2009, at 10:19 AM, Laurent Fabre wrote: Hi, Can I

[Freeswitch-users] Gateway Params

Hi, Can I enclose the params of the directory gateway element in ? I mean I sure can but is the parser going to choke on it or not ? Thanks in advance, -- Laurent FABRE Directeur général 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61

Re: [Freeswitch-users] Conference and socket outbound

is it possible to define a profile and its params for a conference dynamically over socket outbound? in the moment, if we want to have multiple profiles for different clients, we (have to) setup a profile in the conference.conf - otherwise we get an error in fs. because we have multipple fs-server

Re: [Freeswitch-users] voicemail web interface

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, yes, I expected that. I enhanced the voicemail code, so it looks for the file suffix (mp3 or wav) and do a embed tag which calls my quicktime plugin. I know QT isn't available for linux ... :/ Disadvantage: every voicemail file is loaded when web

Re: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches

That was the change i checked into trunk to allow app::arg as well as apparg that doesn't work for you? When i said update it was down to the minute i sent the email that the change was added. On Fri, Jan 23, 2009 at 4:03 AM, Sias Mey wrote: > Woot greater win. > > Thanks you so much for that

Re: [Freeswitch-users] freeswitch memory leak issue

Make sure your mysql lib is the right one, you need the re entrant version look for _r We have seen people getting leaks in mysql in the past in several applications when not using the correct lib. There is not much else we can do to keep you from having a leak in Java and/or mysql client lib. If

Re: [Freeswitch-users] voicemail web interface

in your voicemail prefs you have to change the format to mp3 to be able to play them on the web interface. in voicemail.conf.xml (you must have mod_shout loaded for this option) You also may have to change the recording rate to 11025 I know you do with gmail's player but try it without this se

Re: [Freeswitch-users] mod_g729

Kristian Kielhofner wrote: > On Thu, Jan 22, 2009 at 7:30 PM, Steve Underwood wrote: > >> Depends what you are after. Speex offers the quality of G.729 at around >> the same processing load. However, nobody seems to want to pay for the >> processing load of G.729. Almost everything uses G.729A.

Re: [Freeswitch-users] mod_g729

Spacibah Balshoi When are you making g723 for the Russians? > I wrote it becouse in Russia * is very popular. And it have g729. > > I want to make popular FS in my country. We have not patent issues, but I > more > like speex and celt - it's better in my opinion - 8 kHz is past century! It's

Re: [Freeswitch-users] mod_radius_cdr questions and thoughts

the first one would not appear in the mailing list 2 hours after I had sent it so I figured there was something wrong and I resent it Brian West wrote: Did you mean to send this twice? /b On Jan 22, 2009, at 8:52 AM, Apostolos Pantsiopoulos wrote: I am trying to implement a radius based s

Re: [Freeswitch-users] voicemail web interface

I was having the same issue :) I'm glad you found the workaround. -- Laurent FABRE Directeur général 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fa...@kirranet.com -Message d'origine- De : freeswitch-users-boun...@lis

Re: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches

Woot greater win. Thanks you so much for that pointer. although i did have to change the dialplan line to (space between jsapi and foo.js instead of ::) and im not sure if the api.js file actually made any difference.. but it did point me in the right direction. On Fri, Jan 23, 2009 at 11:50:3

Re: [Freeswitch-users] mod_radius_cdr questions and thoughts

Did you mean to send this twice? /b On Jan 22, 2009, at 8:52 AM, Apostolos Pantsiopoulos wrote: > I am trying to implement a radius based solution > using FS. I have seen that the mod_radius_cdr module > is actively maintained. so I have a few questions/remarks : > > 1) When I place a call and m

Re: [Freeswitch-users] voicemail web interface

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, Update to this: I found that slim.swf only plays mp3 files. wav files, which I have are ignored without an error message. renaming .wav to .mp3 doesn't help. slim.swf sends only then a http request, when the file to be loaded has a mp3 suffix.

Re: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches

Wait sory ignore my previous reply... I only just realized you were actually routing through the javascript xml_rpc module. and I didnt actually have the api.js file in my scripts dir. let me see what this does before you worry about it any more ;-) On Thu, Jan 22, 2009 at 04:25:54PM -0600, Antho

Re: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches

Hmm ok... updated to the latest SVN and tried your suggsestion however all I can see happening in the console is 2009-01-23 11:35:44 [ERR] hangup.js:2 mod_spidermonkey() ReferenceError: request is not defined (obviously I renamed foo.js to hangup here) thanks again for the help. On Thu, Jan 22,

Re: [Freeswitch-users] voicemail web interface

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, well, was all my fault ... I was logged in with xml rpc user (freeswitch/work). Both urls working the same way on my side: entered in both pages simply extension number + vm-password. Both pages results in the same output. Very very impressiv