Re: [Freeswitch-users] Re-2: Ruby and ESL help

Hi guys, It's me again, does anyone knows why this doesn't work? require 'rubygems' require 'eventmachine' require 'ESL' EventMachine.run { con = EventMachine::start_server "127.0.0.1", 8084 do fd = con.to_i esl = ESL::ESLconnection.new(fd)

[Freeswitch-users] Amazon EC2 no audio

Hello all, Help a n00b out.I have been trying to get an instance of FreeSwitch running up in the Amazon EC2 cloud. I have successfully gotten the package built following the wiki and archives of this list. I can get x-lite to register with the switch and it will set up calls o

Re: [Freeswitch-users] Inboud Call Queue

See this: http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote: Thanks Guys From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Anthony Minessale Sent: Wednesday

Re: [Freeswitch-users] Busy tone and text message configuration

> Date: Wed, 6 May 2009 09:47:42 -0400 > From: intralan...@freeswitch.org > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] Busy tone and text message configuration > > chenexyee wrote: > > > > 1. user A is in conversation with user B, and at this time, a incoming

Re: [Freeswitch-users] DTMF recognition flaky

Hi Jay, Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk from a couple of days ago. We handle around 100,000 calls/day via FS) That said, I've found it depends on your SIP trunk provider.That doesn't mean to say there isn't a problem; it's just that I haven't com

Re: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony

On Wed, May 6, 2009 at 11:35 AM, Giovanni Maruzzelli wrote: > On Wed, May 6, 2009 at 8:23 PM, Michael Collins > wrote: > > FYI, > > I made a comment on Dave's blog extolling the virtues of FS and I > mentioned > > Skype support. I didn't specifically mention mod_skypiax but I didn't > > specifica

Re: [Freeswitch-users] DTMF recognition flaky

I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | aus...@signal15.com

Re: [Freeswitch-users] DTMF recognition flaky

Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of

[Freeswitch-users] DTMF recognition flaky

Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just

Re: [Freeswitch-users] Interesting Blog About HD Telephony

Well its not so much which codecs but that the codecs can do 16k... Currently FreesWITCH supports d...@16k, sp...@16k, G722 and g72...@16k But as a bonus: We can also do sp...@32k, g722...@32k, c...@32k and 48k /b On May 6, 2009, at 1:38 PM, Paul wrote: Are the most currently ratified HD

Re: [Freeswitch-users] Interesting Blog About HD Telephony

Are the most currently ratified HD Voice codecs G.722 and G.722.1? I haven't heard very much about HD Voice at all until you just brought it up. From: Michael Collins To: "freeswitch-users@lists.freeswitch.org" ; freeswitch-...@lists.freeswitch.org Sent: Wed

Re: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony

On Wed, May 6, 2009 at 8:23 PM, Michael Collins wrote: > FYI, > I made a comment on Dave's blog extolling the virtues of FS and I mentioned > Skype support. I didn't specifically mention mod_skypiax but I didn't > specifically mention any mods. > I was suggesting to put mod_skypiax in the http://

Re: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony

On Wed, May 6, 2009 at 10:00 AM, Giovanni Maruzzelli wrote: > Ciao Michael, > if you like, you can add that using mod_skypiax you have native hd > skype->FS and FS->Skype (no hardware needed) :-) > > FYI, I made a comment on Dave's blog extolling the virtues of FS and I mentioned Skype support. I

Re: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony

Ciao Michael, if you like, you can add that using mod_skypiax you have native hd skype->FS and FS->Skype (no hardware needed) :-) On Wed, May 6, 2009 at 6:36 PM, Michael Collins wrote: > FYI, > > Here's a nice story for you all to check out. Please check it out and pass > it on. > > -Michael >

[Freeswitch-users] Interesting Blog About HD Telephony

FYI, Here's a nice story for you all to check out. Please check it out and pass it on. -Michael ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/fre

Re: [Freeswitch-users] Sphinx 4 Integration...

Yes you are right, it uses SIP for signaling and RTP for media transport. But does it make any difference because in its documentation they have stated that they support Asterisk and any IP PBX that supports SIP and RTP. I haven't tried it with FS yet but it worked with Xlite using SIP. For some r

Re: [Freeswitch-users] Sphinx 4 Integration...

You're a bit mistaken on the MRCP 2.0 supporting SIP.. it uses SIP for signaling and RTP for media transport. ... that however doesn't mean you can just call it via SIP and it work. They put a little bit more goop on top of that! /b On May 6, 2009, at 9:58 AM, Moiz Chinoy wrote: Hi All,

[Freeswitch-users] Sphinx 4 Integration...

Hi All, Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0 and integrated Sphinx 4. Since MRCP 2.0 supports SIP for communication, it can easily be integrated with FS! Although it is implemented in Java, VoiceXML is also supported! -- Regards, Moiz Chinoy. _

Re: [Freeswitch-users] Inboud Call Queue

Thanks, so quick. Actually I had submitted another version of patch which added a channel var fifo_caller_exit_to_orbit which make the caller possible to exit to the orbit_exten other than hangup the caller when the caller enter the fifo_caller_exit_key. I use this to guide the caller to an

Re: [Freeswitch-users] Inboud Call Queue

Thanks Guys _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, May 06, 2009 3:57 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue

Re: [Freeswitch-users] Busy tone and text message configuration

The exact same way you use it for outbound... just use limit before you call the user in your dial plan. An inbound call to a user is nothing more than an outbound call from FreeSWITCH to the user. /b On May 6, 2009, at 9:11 AM, dujinfang wrote: I'm also finding a way to limit only one ca

Re: [Freeswitch-users] Busy tone and text message configuration

On May 6, 2009, at 9:47 PM, Raymond Chandler wrote: > chenexyee wrote: >> >> 1. user A is in conversation with user B, and at this time, a >> incoming >> call from user C comes to A, in this case, I want freeswitch to play >> busytone to C, how to configure? > you could use the limit app (mod_l

[Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre7 Now Available

FYI, Please update your installations as soon as possible. More information on this update is available here . Thanks for all of your feedback - please keep it coming and join us on IRC if you have any questions about the newest version. -Michael S Collins htt

Re: [Freeswitch-users] Inboud Call Queue

I worked on the patch and added it to trunk rev 13240 On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: > The patch haven't been merged into trunk. It should be as easy as execute > the following command in the FS source code root dir: > patch < /tmp/the_patch_file_name.diff > > I will post an e

Re: [Freeswitch-users] Busy tone and text message configuration

chenexyee wrote: > > 1. user A is in conversation with user B, and at this time, a incoming > call from user C comes to A, in this case, I want freeswitch to play > busytone to C, how to configure? you could use the limit app (mod_limit) to limit A's number of calls to 1, then play the busy sound w

Re: [Freeswitch-users] Profile reloading

The IP guessing code changed its guess then... /b On May 6, 2009, at 7:57 AM, Jonas Gauffin wrote: No, have never changed the IPs since the server was installed. And have not changed it in FS either. Ok. Will add the parameter. Thanks. Brian West br...@freeswitch.org -- Meet us at ClueCo

Re: [Freeswitch-users] Profile reloading

No, have never changed the IPs since the server was installed. And have not changed it in FS either. Ok. Will add the parameter. Thanks. On Wed, May 6, 2009 at 2:01 PM, Brian West wrote: > add > to conf/autoload_configs/sofia.conf.xml > Did you happen to bind the IP while FS was running? > > /b

Re: [Freeswitch-users] Inboud Call Queue

The patch haven't been merged into trunk. It should be as easy as execute the following command in the FS source code root dir: patch < /tmp/the_patch_file_name.diff I will post an example on the wiki when I finished, hope be soon. On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: Hi Seven, I

Re: [Freeswitch-users] Profile reloading

add to conf/ autoload_configs/sofia.conf.xml Did you happen to bind the IP while FS was running? /b On May 6, 2009, at 1:36 AM, Jonas Gauffin wrote: Hello, I have to static IP:s on my server. FS has been bound to one of them. Yesterday evening I got these log messages: 2009-05-05 19:17:37

Re: [Freeswitch-users] Inboud Call Queue

Hi Seven, I am exactly looking for this functionality. Please let me know when you are finished with new queue manager app. I'll try it in my call center. Regarding Patch: is it already part of SVN trunk? If not then could you help me how to install it, I have no programming background. Many Th