[EMAIL PROTECTED] wrote:
On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote:
From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface
From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface.
I determined this from an RTP debug that showed packets sent to the
phone and packets
From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface.
I determined this from an RTP debug that showed packets sent to the
phone and packets
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I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead
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I have a true leased line (a T1) between the two sites.
What parts do I configure for Asterisk to utilized the link bi-directional?
On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
On Wednesday 28 June 2006 08:48, Jonathan Miller wrote
kind of T1? TDM? Data? What type of signaling are you planning
to use em? There is a lot of information that that question is
lacking for anyone to advise you ...
Jonathan Miller wrote:
I have a true leased line (a T1) between the two sites.
What parts do I configure for Asterisk
a package missing, but don't know what that could be and I'm
not able to find much help on this in the history of the list. Other
issues have been resolved, but I can't find anything about this.
Please help!
Sincerely,
Jonathan Miller
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their products every year (or it least that's what happened with the brand
new gear we've bought from them so far.. i.e. as5200, as5350,
catalyst3500xl). I'd prefer someone else who will provide firmware
fixes/updates without a contract. Say, where's the wiki?
Jonathan Miller
-Original Message
?
Jonathan Miller
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Joe Greco
Sent: Saturday, October 16, 2004 8:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Simple phone question
Continuing on/adding my $0.02 to Joe's reply on this thread
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them). We just really
want a new system that will be expandable for the future.
, will the different
lines flash when there is an incoming call? If so, how is this
configured/controlled in asterisk? Just trying to figure this stuff out.
Any suggestions or help would be greatly appreciated. Thanks again
asterisk!
Jonathan Miller
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