Hey,
There is a way to use the new confbridg without installing the new
version of Asterisk,
I'm new in using asterisk, my is version 1.6.2 of Asterisk.
I would not want just to get into the installation of a new version just
for a one commend which I have very great need.
Thank you for your
Hi all,
I tried making a video SIP call using Asterisk But it didnt workonly
voice call works?
Regards
Faraj Khasib
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Hey,
Did you try google.com for this!
I've done this several times now. Video for one-to-one call works if H264
is supported at both end points. All you need to do is enable video in
sip.conf and set allow=h264 in the sip peers with video capability.
You may need to see if your asterisk has h264
Hi all,
Do you have an idea on the best way on how to implement a system with
multiple Asterisk servers with BLF working in such a way that a peer on one
server can subscribe to another peer on the other server in a seamless
manner? Has anyone set-up a system like this before?
Thanks!
Regards,
Le 16/11/2011 10:23, Faraj Khasib a écrit :
Hi all,
I tried making a video SIP call using Asterisk But it didnt workonly
voice call works?
Hi Faraj,
Asterisk support H261, H263, H263+ and H264. Video calls are working
since at least 1.4 version. You have to activate it by setting
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
Asterisk will be supported for two more years, until July 26, 2013.
I want to test this thing. Any Idea. any free
Yes, Skype was a good thing. R.I.P
On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
Now I did, thank you for ur help and it works :D
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI
[ad...@tootai.net]
Sent: Wednesday, November 16, 2011 5:49 AM
To:
I can tell you that siptosis is workable but the support has been dropped
recently as well.
It is a great program and especially the paid version with trunk builder
i.e. you can have multiple skype instances
On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype
Hello group,
I have this situation:
I have several contexts with a few extensions each one. I need to give every
context a limited quantity of minutes they can use. All the extensions in the
context will share the same bag of minutes. Meaning ext 101 use 1900 mins,
ext 102 60 mins and ext 40
On Wed, Nov 16, 2011 at 10:46 AM, Hans Goossen goos...@planet.com.py wrote:
Hello group,
I have this situation:
I have several contexts with a few extensions each one. I need to give every
context a limited quantity of minutes they can use. All the extensions in the
context will share the
Hello Hans,
On 11-16-2011 14:46, Hans Goossen wrote:
I guess some billing solution can do the trick, but I think it's too much for
that little. I don't need any other feature.
i would create a macro which calls an agi. The agi searches the CDR
table (mine is in sql) and calculates if the
But what is the correct physical setup of a CLEC.
Do you get rack space at a carrier hotel and equipment in there?
Do you get rack space at the local ILEC CO?; which is Verizon here.
What are the types of voice platforms used by CLECs?
Thanks,
--E
-Original Message-
From:
On Wednesday 16 November 2011, Abdul Basit wrote:
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
Asterisk will be supported for two more years, until July
On 11/16/2011 10:30 AM, eherr wrote:
But what is the correct physical setup of a CLEC.
Do you get rack space at a carrier hotel and equipment in there?
Do you get rack space at the local ILEC CO?; which is Verizon here.
What are the types of voice platforms used by CLECs?
Just as a point
On Wed, 16 Nov 2011, A J Stiles wrote:
You would be better off persuading Skype users to transition to something else.
Skype is the absolute antithesis of the whole point of telephony, which is to
connect people together. This includes, implicitly, the ability for
subscribers on one
I would agree, unfortunately.
However, I still see it as a glorified webcam chat and not a telecommunication
device like a SIP/soft phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
On 11/16/2011 10:44 AM, Gordon Henderson wrote:
The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my
wife next door and it doesn't use up any of my own broadband bandwidth
wheras if I use a hosted SIP service, calls go out come back in again.
Skype also seems to be able to run
On 11/16/2011 02:17 AM, Eyal wrote:
Hey,
There is a way to use the new confbridg without installing the new
version of Asterisk,
I'm new in using asterisk, my is version 1.6.2 of Asterisk.
I would not want just to get into the installation of a new version just
for a one commend which I have
On 11/16/2011 04:18 AM, Ronald Cepres wrote:
Hi all,
Do you have an idea on the best way on how to implement a system with
multiple Asterisk servers with BLF working in such a way that a peer on
one server can subscribe to another peer on the other server in a
seamless manner? Has anyone set-up
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote:
As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE
NAT traversal mechanism, this will start happening for regular SIP calls as
well. This *should* already happen with the Blink softphone, for
On 11/16/2011 10:30 AM, eherr wrote:
But what is the correct physical setup of a CLEC.
There is no correct physical setup. The setups vary as much as
anything else does, and are shaped mainly by the purpose of the CLEC
and the range of products it provides.
Do you get rack space at a
On the polycom soundpoint ip 650 six line phone:
Say I have 4 lines on hold, is there way to tell who I put on hold.
I cannot see the caller ID of the other lines, only the last line I placed on
hold.
Thanks,
--E
--
I'd say try a2billing- thats abit of an overkill for just this
functionality but you'll get lot or options to play with there.
On Wed, Nov 16, 2011 at 7:02 PM, ad...@3a.hu wrote:
Hello Hans,
On 11-16-2011 14:46, Hans Goossen wrote:
I guess some billing solution can do the trick, but I
On Wednesday 16 November 2011, Gordon Henderson wrote:
On Wed, 16 Nov 2011, A J Stiles wrote:
You would be better off persuading Skype users to transition to something
else.
Sadly, my experience in the SOHO environment is that Skype wins.
[stuff deleted]
And now I'm seeing some of my
Core show channels verbose - if you do asterisk -rx cscv from bash
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 12:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
When you perform an attended transfer, the extension of the person transferring
is displayed to the co-worker.
Can I override the caller ID to display the caller's callerID during a blind
transfer?
Thanks,
--E
--
_
--
Thanks for the response.
What you described is for the CLI.
I am asking is there a way on the phone itself or is there a phone that does
have this capability.
Thanks,
--E
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Upgrade to 10.0 - this isn't available in any of the 1.X flavors because
they had to re-invent the background stuff for it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 12:50 PM
To:
You might be able to use hints/buddies.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 12:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users]
You can do this with a global variable and a dedicated context.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind
Sent: Wednesday, November 16, 2011 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had to
So there is no way to do these with programming.
For instance, setting a variable in the DB and grabbing it to override the
field when transferring?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
On Wednesday 16 November 2011, Hans Goossen wrote:
Hello group,
I have this situation:
I have several contexts with a few extensions each one. I need to give
every context a limited quantity of minutes they can use. All the
extensions in the context will share the same bag of minutes.
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID
during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had
If you store the Global in DB and read it back from DB, it can persist
across reboots and reloads.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Wednesday, November 16, 2011 1:13 PM
To:
Unfortunately, I only have 1.4.26 installed.
What's the next stable version?
Should I go to 1.6, 1.8, or 10
Thanks,
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday,
On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote:
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID during a blind
transfer?
Thanks,
--E
I understand and agree.
There is one client who prefers having the attended transfer still display the
original caller ID because some users still just hit
transfer and hangup. The boss has a few times got caught saying What when he
thought it was an internal call but really wasn't.
Have you tried the conferencing button?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 2:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
On Tue, Nov 15, 2011 at 04:53:02PM +, Tony Mountifield wrote:
In article 4ec296b9.8040...@digium.com,
Jason Parker jpar...@digium.com wrote:
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict
Hola,
estoy instalando un ATA HT503 de Grandstream conectado a Asterisk y todo
funciona bastante bien (llamadas en entrada y salientes).
El unico problema que tengo es con el colgado. Si la llamada entrante va al
buzón de voz y la persona cuelga... el canal queda abierto.
Será que alguien
Hi List,
I have a Polycom FX video unit and I'm thinking maybe I can integrate it
on our Asterisk Server to be able to do teleconference and video as well
via Polycom FX.
I already have oh323 configured on my Asterisk box and I just no idea on
how to let them work.Any help please?
Regards,
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