Hello all!
I always prefer to get referrals from fellow professionals, and this is such a
request. I'm looking for the following:
1. Colocation providers in the chicago area to store a small server for the
purpose of setting up a VOIP service (including pstn connection via Digium
cards) for betwe
Your long pause complaint is the
timeout on the PAP2 before it thinks you're done dialing. The voicemail issue
sounds like the dialplan on the PAP2, what do you use to connect? if it's a
star-code (*), you need *#. in the plan to pass *+any
numbers
SKM
From: [EMAIL PROTECTED]
Comments inline:
>a vim user myself. I don't use most of what you descvribe below,
>however:
>
>> 1. Syntax Highlighting, and ease of updating that highlighting
>
>Update asterisk.vim
Good idea, primary issue being I'd have to learn vim, but it's looking like a
LOT of people agree with the concep
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Hey all,
First of all, hello again! Been a while since I've posted to the
list, but I've been here lurking and watching ;-)
Anyway, I wanted to pose a general question to the list to see
if it turns up new suggestions for everyone/me.
What is your p
He's right, if you have or plan to purchase a large number of the devices,
VOIPSupply will provide you the tools, that's how I got them.
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, February 22, 2006 12:40 PM
To: Asterisk Use
Do you need the configuration file compiler? Once you have that, it's as easy as
setting up the tftp server, and making sure your devices know to pull configs.
Additionally, you can send a SIP NOTIFY packet to the PAP2/SPA-2002 to force it
to update it's provisioned configuration.
I'd have to figu
That's because DELETE is a reserved
word. The queries asterisk sends need to have ` surrounding the name, not ' or
".
I had the same problem but my
employers at the time didn't want to delve into the code, we just used the
options column
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
Hate to keep asking, but I've not been able to find it covered online or in
docs.
I know you can define multiple domains in the sip.conf, but can you define
multiple realms?
For instance, I use a central server that handles a couple of area codes, and I
would like to be able to have authenticati
s
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Friday, December 30, 2005 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outputting human readable info on a VoIP
call'squality?
On 1
Hello,
Anyone know of a program that can analyse the RTP media stream and then output a
human readable graph or other file? I'd like to be able to show jitter,
difference, and if possible, echoes and other articfacts within a file of some
sort. Ethereal can show you a graph, but cannot save it as
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