search for: scheesman

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

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

...lk 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> activated. and this was tod...

...,Dial(a bunch of SIP extensions) But then every call was answered regardless of CID and the tones were heard. Any ideas? G7LTT/KC2ENI Mark Phillips --__--__-- Message: 9 Subject: RE: [Asterisk-Users] Zapateller issues Date: Mon, 12 Apr 2004 14:54:58 -0500 From: "Sean Cheesman" <scheesman@macarthur-group.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com If I remember correctly (and I could be wrong) I think you have to answer the line first... exten =3D> s,1,Answer exten =3D> s,2,Zapateller(nocallerid) exten =3D> s,3,Privacymana...

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

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.......

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

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

Hi all, I have posted a copy of the 3624 manual on the web. It's 11MB and over 650 pages, so not exactly light reading! You can grab it at http://www.caeveo.com/files/newbridge3624.pdf. Please be kind and save it to your local machine instead of reading it from the web! Thanks! Sean -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

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

sounds like you need to do some reading at the many fine resources available. start at http://www.voip-info.org. Here's a hint for you though.... exten => s,1,Answer exten => s,2,VoicemailMain Barring that, just run 'make samples' which will create a wonderful set of sample config files which will allow you to test the system out pretty thoroughly.... Sean -----Original

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. 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

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

Hi John, Try adding username=5702 and username=5703 to each of the configs in sip.conf. I recall I had this problem with the Grandstreams. -----Original Message----- From: John Coll [mailto:john.coll@csoft.co.uk] Sent: Saturday, January 03, 2004 11:56 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)