[Asterisk-Users] VoicePulse IAX2 lag

2004-02-02 Thread Jonathan Tew
Is anyone else noticing high lag on their voicepulse IAX2 connections? We're seeing 500-600ms lag time when our pings to gw5.voicepulse.com are only 90ms. Our other IAX2 connections to this box are not experiencing anything else like this. Our server has hundreds of megs of free ram and is v

Re: [Asterisk-Users] Introducing Firefly

2004-01-29 Thread Jonathan Tew
Just tried loading this up. My mouse was jumping all over the place for some reason. Have you guys tried this software on multi-monitor workstations? Samuel Jimenez wrote: Nice!! Have just tried it a bit, seems cool... Congrats!!! Will test it

Re: [Asterisk-Users] prepaid app

2003-12-26 Thread Jonathan Tew
What if his application is a stub that communicates to another process via TCP/IP. That other process can be closed source without any problem. First off he cannot distribute his C API based app without 1) releasing it GPL or 2) paying Digium for a non-gpl licnese. ___

[Asterisk-Users] Short in my X100P. Is it broke?

2003-12-24 Thread Jonathan Tew
At my home office I have a X100P card in a server that I've been using for testing. The machine it is in is connected to a HP fax machine and then to the wall outlet. This morning the SBC installer showed up at my house for the ADSL install on that line. He said they detected a short. So he

Re: [Asterisk-Users] Asterisk + CRM

2003-12-23 Thread Jonathan Tew
We're starting to integrate * with our customer service software. Basically we're pulling off events from the management interface. We're also making some small patches to the code to deliver more events about the channel variables, etc. Anton Yurchenko wrote: Hello, Anyone aware of any CR

Re: [Asterisk-Users] sip show peers - disappearing

2003-12-22 Thread Jonathan Tew
I think we've having some luck with this setting. Of course we had to crank it up higher so that it didn't consider the clients LAGGED. When the clients were LAGGED they couldn't receive any calls for some reason. So like a setting of 200ms seems to work fine for everyone. Eric Wieling wrote

[Asterisk-Users] sip show peers - disappearing

2003-12-22 Thread Jonathan Tew
We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in "sip show peers" with the

Re: [Asterisk-Users] Garbled VoiceMail

2003-12-13 Thread Jonathan Tew
Burak, Try connecting to your * server with a SIP phone like X-Lite or an IAX phone like DIAX. Do you get the same results with those phones too? Jonathan Burak Balasaygun wrote: Hi, I just got started with asterisk and am having a problem with voice quality. When connecting via either a G

[Asterisk-Users] Streaming Hold Music

2003-12-11 Thread Jonathan Tew
Is it possible to play live streams of audio for the hold music instead of just MP3 files? Like some online radio stations or something? Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] app_queue bug with call transfer

2003-12-10 Thread Jonathan Tew
Kevin, I've looked at the source of app_queue.c and can see if the logic for the * key, but nothing related to # in the code. Am I missing something? Thanks, Jonathan Kevin Bockman wrote: --- Jonathan Tew <[EMAIL PROTECTED]> wrote: We've got the app_queue configured to suppos

[Asterisk-Users] app_queue bug with call transfer

2003-12-09 Thread Jonathan Tew
We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the que

Re: [Asterisk-Users] Problems with voicepulse.com

2003-12-08 Thread Jonathan Tew
Did you make sure to put the "in-" in front of your register => in-login:[EMAIL PROTECTED] Ernest W. Lessenger wrote: At 01:32 PM 12/8/2003, you wrote: Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound

[Asterisk-Users] Echo X100P and X-Lite SIP Phone

2003-12-05 Thread Jonathan Tew
We're using an X100P card to connect to the SBC PSTN and have a SIP client (X-Lite) configured. I'm using a headset (Plantronics 20) to connect to my computer. When I first make a call to someone over the PSTN I can really hear myself echo bad in my headset. As the call goes on it gets bette

[Asterisk-Users] Voice Pulse Account Management Down?

2003-12-05 Thread Jonathan Tew
This is a little off topic, but this is one of the greatest concentrations of potential VoicePulse customers I know of. We signed up with an account last night just to do some testing. We tested the outbound functionality of calling through VoicePulse from * and it worked just fine. We then

Re: [Asterisk-Users] x100p/hangup detection issues?

2003-12-04 Thread Jonathan Tew
Pat, We're testing with an X100P card. When the caller on the POTS line hangs up it never causes our IAX phones (DIAX in this case) to hang up. Curious what you find out. Thanks, Jonathan Patrick Cantwell wrote: Hi.. I've got an asterisk setup with an X100P card installed.. I'm noticing

Re: [Asterisk-Users] voip-info.org is a great Resource ..BUT

2003-12-04 Thread Jonathan Tew
Being farely new to the Asterisk scene and searching for documentation I was wondering why the asterisk.org site didn't run a wiki. There isn't anything as good as a wiki for collecting collaborative documentation. Over time someone might want to convert the knowledge contained in the wik

[Asterisk-Users] Application API

2003-12-03 Thread Jonathan Tew
Asterisk Users, Does anyone know the URL for the application API for asterisk? I haven't been able to find any documentation on it. Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-us