search for: cheesman

have you set up the db schema? and have you entered any sip data into the db? Sean -----Original Message----- From: Chandra [mailto:chandra@digital.com.np] Sent: Tue 1/6/2004 10:57 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] no results. i have been working with the retrieve_sip_conf_from_mysql.pl file and i have set everything as required. but when i

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

Hy Guys, I''m reading a book called "UML Components - A Simple Process for Specifying Component Based Software" (by John Cheesman and Jonh Daniels) and i was wandering if this kind of development is possible on Ruby and Rails as well. Has anyone tried to develop in rails using a Component-Based development approach? Thanks in advance, and sorry if it is a stupid question. :) -- Posted via http://www.ruby-forum.com/. --~-...

...ge in DB (James H. Cloos Jr.) 4. Voicemail config from database (AJ Grinnell) 5. oob to inband dtmf over rtp (James H. Cloos Jr.) 6. OT appologies to list (Linus Surguy) 7. Re: OT appologies to list (Brian Cuthie) 8. Zapateller issues (Mark Phillips) 9. RE: Zapateller issues (Sean Cheesman) 10. RE: Zapateller issues (Darrin Johnson) 11. Re: Re: Voicemail storage in DB (James H. Thompson) 12. Re[2]: [Asterisk-Users] OT appologies to list (Stephen Karrington) --__--__-- Message: 1 Date: Mon, 12 Apr 2004 13:08:06 -0500 From: Eric Wieling <eric@fnords.org> To: asterisk-user...

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

ok, I've got yet another newbie question. My extensions.conf is getting rather longish and I'm getting dizzy moving back and forth editing this thing. Can I use the include command to include a file in order to break extensions.conf up into more manageable pieces ? Is breaking up the extension.conf file an OK thing to do ? Maybe something like this: include

Greetings, Below is part of the contents of my extensions.conf file. exten => s,1,Wait,1 ; Wait a second before answering. exten => s,2,Answer exten => s,3,ResponseTimeout,10 ; Set the amount of time the user ; has to make a selection. exten => s,4,DigitTimeout,5

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -----Original Message----- From: mikeu [mailto:mikeu@execpc.com] Sent: Saturday, January 10, 2004 12:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chagres

...couple times to talk to them about service. How much it costs, JB> how it works, etc. Just common stuff you might find on a website. I left a JB> message and nobody returned my call; I went with voicepulse instead. JB> John JB> ----- Original Message ----- JB> From: "Sean Cheesman" <scheesman@macarthur-group.com> JB> To: <asterisk-users@lists.digium.com> JB> Sent: Sunday, January 25, 2004 9:49 PM JB> Subject: RE: [Asterisk-Users] Has Nufone gone belly-up JB> funny... I got an immediate response, and within 1 hour had my account JB> activate...

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

After searching the archives for a while, I couldn't find any easy way to get everything loaded on startup. So I decided to take a stab at writing some notes on what I've found. If everyone chips in, maybe we can make that part easier for new users! Both the Zaptel and Asterisk packages have a make option called config (make config). This option adds an entry to your init.d directory

voicemail notification? -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Wednesday, December 24, 2003 12:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream 102 flashing display The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight

my biggest concern about defaulting the context to anything at all besides [default] is that you then have to remember to configure the voicemail.conf with the corresponding contexts. as it stands, you have the ability to do just that, but you don't have to. if you have several hundred extensions broken out by dozens of contexts, it might not make sense to force the voicemail.conf to follow