Dear Sir/ Madam,
My client is using Asterisk based telephony system. He has 300 concurrent
users with two PRI Channels.
He is requesting to add the intercom feature in his phones. Can you assist
me please doing that?
I would appreciate your kind responds.
Best Wishes,
Dania
You asked this question before and there was an already answer on September 28.
jg
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Dear JG,
Thank you for following up with me.
Kindly note that I asked about the capability of the phones and now I am
asking about the way I can do it to my client's phones, because he is asking
for a demonstration.
Best Wishes,
Dania Abu Asi
Sales Executive Engineer
Future Trends
It can't be done in analog phones.
On 07-Oct-2014 1:54 PM, Dania Asi da...@futuretrendsest.com wrote:
Dear JG,
Thank you for following up with me.
Kindly note that I asked about the capability of the phones and now I am
asking about the way I can do it to my client's phones, because he is
On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote:
Kindly note that I asked about the capability of the phones and now I am
asking about the way I can do it to my client's phones, because he is asking
for a demonstration.
Yet you’ve not even told us the phones in use. You
Then this may be the wrong forum. Intercom is also a bit vague---there are a couple of different
options. Have a look at: http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
You might just have to set the auto-answer feature of a phone, but this would
be phone specific.
jg
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Dear Mr. Mitual,
Kindly check the attached mail where Mr. JG confirmed to me that is possible
and I already informed my client of that.
Dear Mr. Steve,
I am not expecting a mailing list to do any work for me. All I was asking is
for you to guide me because this is the first time we deal with
Hi,
Thanks Matthew for trying to reproduce the problem, I appreciate your
efforts very much.
There must be something off in my setup in one way or another. I could just
discard this server and build a new one, but I think it's not good practice
to leave a problem unsolved, so I'll continue
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck.
JG confirmed that it is possible, but it has not been defined.
Without knowing what kind of instruments you are using, a possible it
would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.
--Don
-Original
Olli Heiskanen wrote:
Hi,
Thanks Matthew for trying to reproduce the problem, I appreciate your
efforts very much.
There must be something off in my setup in one way or another. I could
just discard this server and build a new one, but I think it's not good
practice to leave a problem
James Lamanna wrote:
Hi Matt,
So this actually works (haven't had a chance to try it)?
SET VARIABLE CHANNEL(musicclass) default
Because musicclass is piece of channel information.
Referencing ${musicclass} is not the same thing.
It should indeed work, yes.
--
Joshua Colp
Digium, Inc. |
Hi Joshua,
Excellent! I didn't even remember to consider newer versions of asterisk as
11.11 was the latest one when I started building on. I have had libuuid and
libuuid-devel installed the whole time, but perhaps 11.11 just did not see
it there. I just installed 11.13 and it works perfectly.
On Tue, 2014-10-07 at 08:37 -0500, Don Kelly wrote:
JG confirmed that it is possible, but it has not been defined.
Without knowing what kind of instruments you are using, a possible it
would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call
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