ubject: Re: [Freeswitch-users] XMPP (mod_dingaling) for
Events/Messaging
From: Michael Jerris m...@jerris.com
Date: Fri, September 25, 2009 7:22 pm
To: freeswitch-users@lists.freeswitch.org
see chat_send api command and api_hangup_hook. In combination that might work.MikeOn Sep 25, 2009, at 6:07 PM, Pe
Hello all, I was wondering if anyone has used mod_dingaling for messaging rather than voice/video. Specifically, I would like to have FS send an XMPP message to an ActiveMQ server when it records a voicemail. Additionally I would like like to have CDR entries posted into the ActiveMQ server as
I've worked with a few of these commercial systems, so let me provide my perspective. Or just skip to the end for my solution. Disclaimer: I am not an expert with all things mod_fifo can do.
First, I don't like the idea of using the work "skill" because I am one of those in the camp that the work
Actually, getting the 100K is probably the easiest part of the battle. The problem will come in with the strings attached to the 100K. I could persuade my company to give FS devs the money, but there would be conditions, such as:- There would need to be a defined timeline, milestones, and release
I received a similar error trying to compile cepstral on a 64-bit OS. I don't think Cepstral supports 64-bit, as the SDK is 32-bit.-pete
Original Message
Subject: Re: [Freeswitch-users] Troubles build with mod_cepstral
From: Max Bridgewater max.bridgewa...@gmail.com
Date: Sun,
I am also in the process of playing around with FS running inside Xen Virtual Machines with "mirrored" VMs on a second failover system. So far, the initial tests are promising. I can see a 2-3 sec "hiccup" in the net traffic during the live migration of the Xen VM. My calls do not drop, but I will
was our choice. There's a very nice tutorial on their website on how to configure Load Balancing.-pete
Original Message
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
From: "Pete Mueller" p...@privateconnect.com
Date: Sat, August 29, 2009 2:25 pm
To:
dbalancing
From: Steve Underwood ste...@coppice.org
Date: Sat, August 29, 2009 8:17 pm
To: freeswitch-users@lists.freeswitch.org
This sounds like so many "redundancy" projects that will probably offer
nothing in the real world.
On 08/30/2009 05:52 AM, Pete Mueller wrote:
I guess I
Good morning all,I realize this is slightly off the FS topic, but I am wondering if anyone out there has experience with software packages designed for the transcription of voicemails to text. I've used pocketsphinx with FS to handle IVR menus, but now have the task of figuring out how to convert
I apologize, I should have been more clear. We will be using humans to scan the translated results. But we are looking for a system to perform the "first pass" on the audio to hopefully help the human type less.Although the question has been raised if it's faster to have a human just transcribe
Yes, you can use the stream global object. example: local api = freeswitch.API(); local reply = api:execute("originate", someRoute); if (reply) then stream:write("RESULT: " .. reply .. "\n"); else stream:write("ERROR") end
Original Message
Subject: [Freeswitch-users] Lua Script
I believe you need to set LUA_PATH, here's more information:http://www.lua.org/pil/8.1.html-pete
Original Message
Subject: [Freeswitch-users] Lua on Windows and additional modules
From: Vladimir Rodionov vladrodio...@gmail.com
Date: Thu, August 06, 2009 5:55 pm
To:
Disclaimer: I'm not familiar with all the mods of FS, There may be one that does this already. There are probably many ways to do this, I am just offering one that works well for me.Item #1 - Findout the callee #. "destination_number" can be set to several different things based on the gateway
Message
Subject: Re: [Freeswitch-users] Confusing handling of incoming calls
From: Rupa Schomaker r...@rupa.com
Date: Wed, July 22, 2009 2:12 am
To: freeswitch-users@lists.freeswitch.org
On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.com wrote: My goal is:0) figure out why
From: Rupa Schomaker r...@rupa.com Date: Wed, July 22, 2009 2:12 am To: freeswitch-users@lists.freeswitch.org On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.com wrote: My goal is:0) figure out why the bandwidth gateway is being processed as "internal" (th
I have two different gateways setup on my server. One with FlowRoute (which uses SIP REGISTER) and one with bandwidth.com (which does not). I can send and receive from both gateways, but the dialplan processing seems to be confused. Calls from bandwidth do not respect the "Extension" param in the
Quick question ya'll:
Is there a way to detect DTMF during bridged conversation? Or do I have to
use a conference for that?
Scenario: Caller A dials in and is bridged with caller B (in my case via lua
- session:execute(bridge, some_route)) If at any time during the
conversation caller B
Hey guys. Has anyone else experienced the inability to detect/receive DTMF
tones? Just yesterday I had about 4-5 hours where One of my IVR scripts
would not detect 1, 2 or 3, but detected the other digits perfectly. If I
removed the sound file that was playing, and substituted silence it
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