Displaying 20 results from an estimated 10000 matches similar to: "keeping interaction terms"
2007 Jun 15
0
No subject
using Asterisk.
=20
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
=20
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to
2009 Jan 16
0
No subject
1. a clause in iphone Developpers agreement that forbid applications runnin=
g in background,
2. lack of sip clients.
Now it seems skype is available on iphones.
Has someone tried it ?
Along new skype capabilities in Asterisk, could it be used to hook iphones =
to Asterisk for both inbound and outbound calls ?
Regards
--_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0242cworksmailcwo_
2009 Jul 20
0
No subject
might be your best bet to get the information you want. I'd look at
voip-info.org for information.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 16, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to list ongoing calls
2011 Jan 10
0
No subject
and Asterisk is plugging in pseudo ID. Is that correct?
It seems to me that Asterisk should simply say "no caller ID" or "No ID" or
something besides "Asterisk".
In any case, we are trying to filter them with little success.
When we do a LEN(CALLERID(num) we get "13", when we expect "10"
The call pattern is 1 call followed by a
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run
Apple doesn't accept (for the moment) an application runs in the background=
. So, when Siphon doesn't run, the SIP server of your provider doesn't know=
your iPhone."
--_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_
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2007 Jun 15
0
No subject
using Asterisk.
=20
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
=20
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to
2007 Jul 12
0
No subject
display, accelerometer/motion sensor being the first 4 for release
(though 81 have been mocked up so far).
The long term concept is if you want a 'weather station with live video
feeds and gps location control you can add various modules together to
deliver what you are looking to achieve.
I have high hopes for the concepts, and wish the guys well as it seems
their hearts are in the right
2007 Jul 12
0
No subject
don't have a public facing web page but you are looking for people to
click on but a personalized list of numbers. In order for someone to
access this directory you are going to be asking for a username/password
correct? If so just tie the username to a selection of 'my location'
checkboxes that I tick and then the app remembers this location next
time I log in (eg server side
2011 Jan 10
0
No subject
major undertaking. But since you are using an AGI to control the Queue
command instead of using it from the dialplan, you have more control over
this problem than you realize. For simplicity of illustration, let's say
your AGI simply wants to take a call and send it to the next agent in the
queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the
Polycom transfer from
2009 Jul 20
0
No subject
used Kamalio to "supplement" the features that Asterisk either doesn't
provide or doesn't provide in as nice a form as the OP desired - can't
really speak beyond this as I am not one of them.
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2004 Sep 21
2
SIP termination in Brazil
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
Best regards,
Han
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2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
2009 May 26
2
Domains
Hi,
I'm trying to understand an issue I'm seeing between two Asterisk
servers. I think it has to do with Domain definitions.
Server A), has extension 5550 defined. Has a sip client 2000 defined,
and has guest-invites enabled.
Server B), Dials to server A for any 5550 dialled. Has sip client 2000
and 2001 defined.
If I register at server B as client 2001, and dial 5550 then
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2004 Jul 14
1
Digium X100P card to a brazilian analog line
Hello,
I have a problem with connecting a Digium X100P card to a Brazilian analog
line.
Can somebody help me out with this problem?
My /etc/zaptel.conf is
loadzone=br
defaultzone=br
fxsks=1
My /etc/asterisk/indications.conf
[general]
country=br
[br]
description = Brazil
ringcadance = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion =
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
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2009 Jul 20
0
No subject
I got this notion
monitor-format = wav49
wav49 presents much louder than regular wav and gsm in my experience
--
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 22, 2010 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
2007 Jul 12
0
No subject
community there is a real possibility this may come off so if you have
an interest in this space and want to contribute to the discussion then
this is your opportunity to do so.
=20
I look forward to all opnions on this topic.
=20
The slide deck for the agenda of this call is located here
http://voipusersconference.org/2008-05-09-Slides=20
Cheers,
Dean=20
________________________________
2007 Nov 28
1
Digium TE120P versus Sangoma A101D-X
Hello List,
We purchased a TE120P card from Digium and it works great. The only
problem is that we are still experiencing echo on some calls. I've tried
various echo cancellers (right now we are using OSLEC) and still no
luck.
My question has anyone gone from the TE120P to a Sangoma A101D-X Single
Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference?
Also I called