Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip provider

2008-03-21 Thread Matthew Warren
?action=detailsddomain=cdsportal.netserver=whois.opensrs.net Sure he has 99.9% uptime since he just this purchased site 3/14/2008. Matthew Warren My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008

[asterisk-users] Polycom behind NAT won't register to * server behind ALG

2007-08-21 Thread Matthew Warren
Polycom's were simply not originally built for multi location VoIP. There is no NAT support in the Polycom's. We have several networks, being an ISP, and have found that when transversing one network say 192.168.2.x with the * box on a 192.168.1.x the polycoms were able to communicate however

[asterisk-users] Re: Help: CallerID Name not being sent

2007-03-12 Thread Matthew Warren
It has been my experience when working with PRI's that you have very limited options when dealing with outbound CID. Due to restrictions of 911 most Telco's will have to have the PRI split into Trunk Groups for proper CID delivery. This would work for a situation of sharing one asterisk server

[asterisk-users] RE VoipNow 1.2.0 Beta

2006-08-08 Thread Matthew Warren
Yes it is an addon of Plesk, thats stating the obvious. But while your complaining about people writing stuff to use what are you doing. If your not a developer don't critisize the developers. I see nothing more than you displaying that you are the Vice President of a 2 man consulting firm.

[asterisk-users] Re: Operator Console(s)/Shared Call Appearances

2006-07-25 Thread Matthew Warren
Subject: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk and I'm trying to determine the best way to allow our receptionist to answer certain executives telephone lines. It seems there are probably two

[asterisk-users] re:Simple But important question (for me)

2006-07-20 Thread Matthew Warren
Actually it is a non-commercial solution he needs and you offered a commercial one in a non commercial group. ;-) Cheers Gonzalo there is a small chance of that, seeing that an address validation script is used for just that address validation and is implemented in customer service applications

[asterisk-users] re:Simple But important question (for me)

2006-07-19 Thread Matthew Warren
We build custom scripts for Asterisk. We can build this for you, for reletivly inexpensive. But you will need to contact me thru email at mwarren at procomconsulting dot com .. This is a commercial app you need but requesting on a non commercial group. Matthew Warren

[Asterisk-Users] php-snmp

2006-06-22 Thread Matthew Warren
Has anyone been able to get PHP-SNMP working on an asterisk box. I have downloaded the net-snmp, utils,libs, perl and php-snmp but unable to get the php to work wit it. It works via command line // without php. This is set up on an AAH box. ___

[Asterisk-Users] Sip stuck

2006-06-14 Thread Matthew Warren
I am showing 3 channels stuck 2 in INVITE and 1 in BYE. I have tried reloading sip, reloading extensions, hangup, rereading configs, even changing usernames in sip.conf, rebooting phones and so forth but none are escaping this. Configs show for sip are as follows: [232] username=232 type=friend

[Asterisk-Users] syslog server

2006-06-06 Thread Matthew Warren
Does anyone know a good syslog server to use for grandstream phones? I want to set this up to see what is happening with the grandstreams. Easy and Free preferably. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] Re: syslog server

2006-06-06 Thread Matthew Warren
hmmm... I am a huge fan of syslog-ng, but the stock syslog on your * system should work well... How do configure this on the asterisk box for the phone and how do you access it? will you be limited to the 2,000 lines of code. The 2,000 lines goes way to quick.

[Asterisk-Users] PHP-AGI help

2006-06-02 Thread Matthew Warren
Can someone help me with this AGI script to send an email. It just isn't working. The file is being called in the dialplan and is saved as em.agi but it isn't sending the email. #!/usr/bin/php4 -q ?php ob_implicit_flush(true); set_time_limit(6); $in = fopen(php://stdin,r); $stdlog =

[Asterisk-Users] email a message

2006-06-01 Thread Matthew Warren
Is there away to set in the dialplan to have asterisk send an email to someone letting them know a caller called with Timedate caller Id and so forth. My system is set to call in say extension 100 goes to context 1 [1] exten = s,1, answer exten = s,2, Background(file) exten = _,1,

[Asterisk-Users] Audio problems 50% of the time. (kurt x)

2006-05-17 Thread Matthew Warren
Sounds like you have to much NAT interference Matt -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.6.0/341 - Release Date: 5/16/2006 ___ --Bandwidth and Colocation provided by

[Asterisk-Users] RE: GrandStream GXP-2000

2006-04-27 Thread Matthew Warren
Grandstream setting needs to be the 3rd radio button (cannot remember the label) Asterisk needs to be the rfc2833 This is the standard setup for Grandstream phones to work with Asterisk. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus