?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
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
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
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.
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
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
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
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.
___
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
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.
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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.
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 =
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,
Sounds like you have to much NAT interference
Matt
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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.
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