waiting indication. Can this be done from the dialplan?
THanks
BEN BROWN
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I have configured the voicemail.conf file as per the wiki to email
voicemails as an attachment. I cannot find any instructions/locations to
set the outgoing server login information. Furthermore, I can get no
emails from asterisk. Can anyone point me to the next step to setup the
attachment of
Is this standard in the industry? My local telco in the US wants
$1050/month for T1 (not PRI!) I can buy 24 POTS lines for $840/month.
Gotta love small towns!
BEN
Sean Cook wrote:
Not sure about where you are but 16 pots lines generally run about $25-$30 /
month = $480/month. For about
these days
depending on where you are, but insist on PRI unless the service
central office
can not provide it.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Ben Brown
Sent: Thursday,
September 08, 2005
9:41 PM
To: Asterisk Users Mailing List - Non-Commercial
Preparing to order a T1 (not PRI) for our asterisk box. The telco has
offered me several options that I am not sure of. Which would be best
for use with asterisk? The box has the Digium card in it, BTW.
1. Dial Tone - No, Yes - Precise, Yes - SCC
2. Framing - SF, ESF
3. Line Coding - AMI, B8ZS
line coding) . If you have it avaible to you, Signaling
type should be PRI. The rest of your numbers 4-7 are in the PRI
signaling. No sound differences in digital. Caller ID is very
important. PRI signaling is very easy to set up with Asterisk.
Ben Brown wrote:
Preparing to order a T1 (not PRI
Thanks for the replys. I'm
convinced. PRI it is.
Peter Svensson wrote:
On Mon, 5 Sep 2005, Ben Brown wrote:
So the only difference with PRI is caller ID? What I am trying to
determine is if the PRI has enough advantages to give up the voice
channel used by the D channel
Any Particular
recommendations on PRI protocol? I can chose from 4ESS, 5ESS, and NI1
Thanks for all your help!
BEN
Ben Brown wrote:
Thanks for the replys. I'm
convinced. PRI it is.
Peter Svensson wrote:
On Mon, 5 Sep 2005, Ben Brown wrote:
So the only
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Ben Brown
Sent: Friday, September 02, 2005 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] random beeps in MeetMe
I have 3 users in a meetme conference. 2 of them are monitor only. I get
I have 3 users in a meetme conference. 2 of them are monitor only. I get
a random beep in the audio during the conference. There appears to be no
pattern. The 2 monitors are SIP softphones and the third is a POTS line
on an XP100 card. disconnecting either of the monitors does not resolve
the
what you mean in regard to
'monitorin' and 'placing the others on hold'. Normally you 'place
someone on hold' after you have spoken to them - so I guess I am not
clear on a few points.
Mark
On 8/28/05, Ben Brown [EMAIL PROTECTED] wrote:
I am looking for a single soft phone application
to use my method, thus I am
seeking a 24 line softphone.
Thanks for the thoughts
BEN
Steve Edwards wrote:
On Sun, 28 Aug 2005, Kevin P. Fleming wrote:
Ben Brown wrote:
I suppose if there was just a way to
monitor the 24 conferences on request, then the participation could be
ac
I am looking for a single soft phone application that is capable of a
minimum of 24 concurrent lines. Suffice to say that I have a somewhat
unique application here, and I would like all connections active all the
time. I want to be able to switch between them for monitoring purposes,
placing
Is there any way to set the maximum length of the voicemail based upon
which context the mailbox is in? I have only found the global setting.
Thanks
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