[Freeswitch-users] Problem with Freeswitch capturing DTMF

2008-12-02 Thread Keith Wood
Hi, I am wondering if I am the only one getting this problem or not. When sending in DTMF to freeswitch, freeswitch is not always capable of capturing all the DTMF being sent. For instance, sending 1000 to freeswitch may end up becoming 100 or 10003 becoming 1003. Am I the only one getting

Re: [Freeswitch-users] Problem with Freeswitch capturing DTMF

2008-12-02 Thread David Knell
Hi Keith, I was just writing a note along similar lines to Mike's. If you need a hand getting a packet capture or interpreting it, drop me a note off-list. Cheers -- Dave We generally are as good as possible on capturing dtmf reliably. If you are seeing dropouts like that I would have

[Freeswitch-users] Question about wrapping libfreeswitch

2008-12-02 Thread Woody Dickson
Hi, I am sorry again for sending another email to the group again. I am working on embedding libfreeswitch to provide better monitoring. The first thing I attempt to do is to run the sample code provided in the website: #include switch.h int main(int argc, char **argv) {

[Freeswitch-users] Wrong # in voicemail

2008-12-02 Thread ccav
My dialplan is pretty simple. I have a single trunk with a vonage softphone DID (1303... we'll call it main) and a virtual DID (1816...) which rings the softphone DID. All incoming calls show up as from softphone DID but the sip_to_user holds the actual number dialed so I can enter the dialplan

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Dennis
we configured mod_shout and are able to record mp3. but if we start to playback the file, it will only be played back to that point, which was recorded, when we started the player. we do this with api uuid_record uuid start /var/www/test.mp3. we are also able to playback a (radio-)stream to an

[Freeswitch-users] Linksys/Cisco SPA400 (ATA 4 Line FXO) Now Documented in Wiki

2008-12-02 Thread Karl Vesterling
Folks; I've just taken the time to document the Sipura, err, Linksys, errr Cisco SPA400 4 line FXO Analog Telephone Adapter in the Wiki. http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo If anyone uses these ATA's and has questions about it let me know and I'll see if I can answer

Re: [Freeswitch-users] Problem with Freeswitch capturing DTMF

2008-12-02 Thread Michael Jerris
We generally are as good as possible on capturing dtmf reliably. If you are seeing dropouts like that I would have to guess that this is a very lossy line. Could you try and look at the packet capture of a call that is missing digits and see if you are indeed dropping a lot of packets.

Re: [Freeswitch-users] libfreeswitch question

2008-12-02 Thread Michael Jerris
On Dec 2, 2008, at 5:55 AM, Woody Dickson wrote: Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling:

[Freeswitch-users] libfreeswitch question

2008-12-02 Thread Woody Dickson
Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [EMAIL PROTECTED] fs]# gcc switchnode.c

Re: [Freeswitch-users] Wrong # in voicemail

2008-12-02 Thread Brian West
Can you show me the full XML for this extension including the regular expression? /b On Dec 2, 2008, at 7:25 AM, ccav wrote: transfer to voicemail is as follows action application=answer/ action application=voicemail data=default $${domain} $2/

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Brian West
Are you on SVN trunk or what rev are you trying to use? /b On Dec 2, 2008, at 7:48 AM, Dennis wrote: it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if we do api uuid_record uuid start shout://user:[EMAIL PROTECTED]:12345/ (and other

Re: [Freeswitch-users] Dialing tone when placing a call with portaudio

2008-12-02 Thread Michael Jerris
What are you calling, sip I assume, this may be a case where the sip signaling is sending a 180 ringing instead of a 183 and we are not generating ringback in that case. Can you please confirm that and test if setting the ringback channel variable before bridge fixes this issue? Mike On

Re: [Freeswitch-users] Console Dialing in Freeswitch

2008-12-02 Thread Michael Jerris
What revision of freeswitch is this? Can you please test this with svn trunk? Mike On Dec 2, 2008, at 2:27 AM, Baskar wrote: Hi, I have updated all the above events you told .It's working fine but when i call extension 1002 from freeswitch console, call is connected to extension 1002,

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Dennis
i am using the latest svn trunk from today. 2008/12/2 Brian West [EMAIL PROTECTED]: Are you on SVN trunk or what rev are you trying to use? /b On Dec 2, 2008, at 7:48 AM, Dennis wrote: it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Brian West
And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at 9:03 AM, Dennis wrote: i am using the latest svn trunk from today. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Dennis
no, not yet. i am still fiddling arround with icecast2. we tried it with someone, who offers radiostreams. perhaps this just works with icecast(2) and shoutcast? 2008/12/2 Brian West [EMAIL PROTECTED]: And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Brian West
icecast2 is a known working server we have talked to before. /b On Dec 2, 2008, at 9:25 AM, Dennis wrote: no, not yet. i am still fiddling arround with icecast2. we tried it with someone, who offers radiostreams. perhaps this just works with icecast(2) and shoutcast?

[Freeswitch-users] Dialing tone when placing a call with portaudio

2008-12-02 Thread Rene Pankratz
Hello, when using mod_portaudio for calling somebody I don't hear anything until the other party answers the call. Is it possible to play a dialing tone when the other party is ringing? Best regards René Pankratz ___ Freeswitch-users mailing list

Re: [Freeswitch-users] Question about wrapping libfreeswitch

2008-12-02 Thread Michael Jerris
I think the api changed a little bit for this. The easiest starting point would be to just clone switch.c and chop out any of the stuff you don't need, it's mostly argument handling code in there. Mike On Dec 2, 2008, at 7:05 AM, Woody Dickson [EMAIL PROTECTED] wrote: Hi, I am sorry

Re: [Freeswitch-users] Problems with Mod_openMRCP

2008-12-02 Thread Anthony Minessale
If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably

Re: [Freeswitch-users] Problems with Mod_openMRCP

2008-12-02 Thread Anthony Minessale
FreeSWITCH has an enterprise scale SIP UA. Not only can it listen on other ports it can listen and work on as many ip:port combos as you want simultaneously each with it's own specific config. If you have an affinity for port 5060 you can always bring up 2 IP on the same box and give one to each

Re: [Freeswitch-users] Listen to a file, while recording?

2008-12-02 Thread Dennis
sorry, problem solved :-) it works very good with icecast2. 2008/12/2 Brian West [EMAIL PROTECTED]: And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at 9:03 AM, Dennis wrote: i am using the latest svn trunk from today.

Re: [Freeswitch-users] Console Dialing in Freeswitch

2008-12-02 Thread Anthony Minessale
from the source tree of FS please type make current when it completes, retest the call. On Tue, Dec 2, 2008 at 5:07 AM, Baskar [EMAIL PROTECTED] wrote: *Hi, This is the svn version i have installed before a month FreeSWITCH Version 1.0.trunk (10130M) * -- *Warm Regards, N.Baskar*

Re: [Freeswitch-users] Windows is slow?

2008-12-02 Thread Carlos Talbot
Have you tried the latest msi build? It's based off svn 10564. Carlos On Sun, Nov 30, 2008 at 11:03 AM, Per Møller [EMAIL PROTECTED] wrote: I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same

Re: [Freeswitch-users] Support for Junghanns duoBRI

2008-12-02 Thread Stefan Knoblich
All HFC-based cards supported by bristuffed Zaptel should work. Stefan Am Monday 01 December 2008 schrieb Michael Jerris: The bri support is still in development, basic calls on ptmp bri do appear to work, although I am not sure with what hardware. Mike On Dec 1, 2008, at 10:26 AM,

Re: [Freeswitch-users] Problems with Mod_openMRCP

2008-12-02 Thread Andrew Gilbert
Mark and David, I am willing to help some with testing here as well, if you need it. Ping me directly or we can get on the IRC. I am on Mac OS, but have readily available vm's with Debian, etc. I also have Prophecy. I have a general interest in an ASR solution as well. Voxeo is great,

Re: [Freeswitch-users] TLS receiving calls

2008-12-02 Thread Anthony Minessale
Naturally, either way is stupid. The whole idea of putting the transport in a uri param is equally stupid to using 2 different protocol names but since SIP is the descendant of http it they decided to stick with the stupidity of http/https and have sip/sips which is almost as if it was designed

Re: [Freeswitch-users] Problems with Mod_openMRCP

2008-12-02 Thread Anthony Minessale
from build root: svn co -r8809 http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox They did seem to express an interest in granting some dev licenses when they realized we took the code out of tree but I have not actually dealt with the issue

[Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Dennis
hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived at the fax feature :-) i am not sure if the docs are up to date or if there was a lot of development in the meantime. therefore i would like to ask, what is possible and what will come in the near

Re: [Freeswitch-users] Problems with Mod_openMRCP

2008-12-02 Thread Anthony Minessale
They contacted us shortly thereafter and asked if we want to have them sell you the license for 50 bucks. hmm, i wonder why i deleted the module. I will tell them that if they give you a developer license you will work on getting it back into trunk. On Tue, Dec 2, 2008 at 11:27 AM, Andrew

Re: [Freeswitch-users] Support for Junghanns duoBRI

2008-12-02 Thread Sergey Kirillov
Cool. Thanks for the answer. All HFC-based cards supported by bristuffed Zaptel should work. Stefan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] TLS receiving calls

2008-12-02 Thread Kristian Kielhofner
On 12/2/08, Anthony Minessale [EMAIL PROTECTED] wrote: Naturally, either way is stupid. Word. The whole idea of putting the transport in a uri param is equally stupid to using 2 different protocol names but since SIP is the descendant of http it they decided to stick with the stupidity of

Re: [Freeswitch-users] TLS receiving calls

2008-12-02 Thread Kristian Kielhofner
On 12/2/08, Anthony Minessale [EMAIL PROTECTED] wrote: We'll schedule a round table with the topic SIP OMFG STFU At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D Heh. I've been trying to make it back these last couple of years. I just might make it in '09! -- Kristian

Re: [Freeswitch-users] Wrong # in voicemail

2008-12-02 Thread ccav
Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous, has been removed. regex is now: ^([01]?)(8162565804)$ Didn't fix the problem but I'm a perfectionist, had to be changed. :D -- View this message in context:

Re: [Freeswitch-users] Windows is slow?

2008-12-02 Thread Per Møller
I checked out the trunk version, and it's still slow. However I found one improvement - it does not crash on shutdown anymore. Could anymore give me some pointers on how to try to debug this on the Windows platform? // Per Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Carlos

Re: [Freeswitch-users] Windows is slow?

2008-12-02 Thread Michael Giagnocavo
Can you do a console loglevel debug, then send all the output around that time? Apart from that, the quickest way might just to attach a debugger, then break all when it pauses and see where the threads are :). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

[Freeswitch-users] Javascript ODBC on Windows

2008-12-02 Thread Joe Bain
Hi all, Is it possible to use mod_spidermonkey_odbc with a Windows installation of FreeSWITCH at the moment? If so does anyone have any pointers? I get: 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect() Connecting ivr_test 2008-12-02 14:23:57 [ERR] switch_odbc.c:160

Re: [Freeswitch-users] Windows is slow?

2008-12-02 Thread Anthony Minessale
is it stun timeout ? do you have one of the ip set to stun:foo ? On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo [EMAIL PROTECTED]wrote: Can you do a console loglevel debug, then send all the output around that time? Apart from that, the quickest way might just to attach a debugger,

Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Michael Collins
Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC On Tue, Dec 2, 2008 at 9:40 AM, Dennis [EMAIL PROTECTED] wrote: hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived

Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Kristian Kielhofner
On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC Can you (or someone) elaborate on this? Maybe the answer really is no, but what about

Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Michael Jerris
T.38 passthrough IS supported, T.38 endpoint and gateway are not yet supported. Mike On Dec 2, 2008, at 4:28 PM, Kristian Kielhofner wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info:

Re: [Freeswitch-users] Javascript ODBC on Windows

2008-12-02 Thread Michael Jerris
Yes, it should work fine. As the error message says it didn't find the data source name you specified. You need to setup your odbc data source on the system Mike On Dec 2, 2008, at 9:29 AM, Joe Bain wrote: Hi all, Is it possible to use mod_spidermonkey_odbc with a Windows

Re: [Freeswitch-users] Wrong # in voicemail

2008-12-02 Thread Brian West
After you set ${dialed_user}=$2 try using ${dialed_user} everywhere instead of $2 just to test. /b On Dec 2, 2008, at 1:29 PM, ccav wrote: Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous, has been removed. regex is now: ^([01]?)(8162565804)$ Didn't fix

Re: [Freeswitch-users] Wrong # in voicemail

2008-12-02 Thread ccav
Made the change, no joy. Do I need to set sip_req_user to the updated DID? Also, I misspoke in my first post, apparently the bridge is NOT going through either. Is there some var/param I can set with $2 so I can see it in the info? -- View this message in context:

Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Michael Collins
On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC

Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?

2008-12-02 Thread Michael Collins
Kristian, Are you on the IRC channel by any chance? -MC (IRC: mercutioviz) On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info:

[Freeswitch-users] Bridging from Event Socket API

2008-12-02 Thread Klaus Teller
Hi Folks, so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. What i want to do is to bridge an existing leg (Unique-ID is known) to a

Re: [Freeswitch-users] Bridging from Event Socket API

2008-12-02 Thread Michael Collins
You probably have several options depending upon your needs. Could you elaborate a bit on what the big picture is? Also, what exactly were you doing when you established the second call leg? Did the second call let get created and a valid uuid assigned, etc.? Just checking. Let us know, MC On