otaro
<[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 betwe
>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 receive
>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 receive
gt; What kind of T1? TDM? Data? What type of signaling are you planning
> to use e&m? 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.
> &g
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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 Mil
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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 o
t happening.
I changed the default RTP ports to be the same as those on the phones,
16384 > 32778
but am still having trouble getting the audio to play on this machine. I
suspect a package missing, but don't know what that could be and I'm
not able to find much help on this in the h
Hi all,
I'm trying to put together a list of gear w/prices to implement an
asterisk system. Does anyone know a good place to buy polycom phones?
Their website isn't much help. Specifically looking for IP500 and IP600
phones. Thanks again!
Sincerely,
Jonathan Mille
Thanks Joe. The polycom phones look pretty good. The only odd thing is
that they make you use a reseller in order to get updates ;(
Sincerely,
Jonathan Miller
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Joe Greco
&g
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
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.
> -Origina
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
than a 1 line phone?
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
>
> &g
phone, 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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