this extension)?
Regards
Faraj Khasib
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asterisk-users
. Fleming
[kpflem...@digium.com]
Sent: Tuesday, November 15, 2011 8:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple SIP endpoint registrations
On 11/15/2011 07:28 AM, Faraj Khasib wrote:
Hi guys,
I want to ask if its possible to make calls using one SIP account
] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, November 15, 2011 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple SIP endpoint registrations
I have phone system and I am connecting Asterisk to it trunk.
Now I want my iphone
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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Subject: Re: [asterisk-users] Does Asterisk Support SIP Video Call ?
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
Hi Asterisks Developers,
I want to learn all about IP telephony and I was wondering If I can participate
in development?
Thank you
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Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e
(email=m...@me.com) but when I check the Message at the target that header is
not there
So I is Askterisk altering the Message and Is there away to include extra
headers for SIP INVITE Message?
Thank u
--
Please guys anybody knows How can I send a unique token to the Receiver at the
Invite call? Is that possible?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com
...@evaristesys.com]
Sent: Sunday, November 27, 2011 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 04:27 PM, Faraj Khasib wrote:
Please guys anybody knows How can I send a unique token to the
Receiver at the Invite call
Balashov
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:19 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers ofINVITE
Message
On 11/27/2011 04:53 PM, Faraj Khasib wrote:
I tried that with my SIP Cleint but the custom
/27/2011 05:25 PM, Faraj Khasib wrote:
Yes, see attached ... Proxy server alter my Test custom header and
delete it, Is there a way to include it in message sent from SIP
Proxy to target?
That would be a proxy configuration issue, wouldn't it?
In principle, the proxy should be passing
Any body knows how I can configure Asterisk SIP to pass all Header Parameters?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Sunday, November 27, 2011 4:50
, and you will see your
header in the outgoing INVITE. Of course this means that your dial plan need
to know which headers to pass.
// T
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: den 28
it will be recieved as Video Call, Can you
plz help me try to solve this problem? Where should I change the Call Media
Type at Asterisks
Regards
Faraj Khasib
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New
Hi All,
I am trying to record Call, but when the call is done I have one file but the
conversation inside it is separate into calls conversation and receiver
its single file but separate recording,
How can I make it mixed together so the conversation will be normal?
Thanx
--
: [asterisk-users] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday
Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non
get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing
I already searched using grep for the monitor word ... It doesn't exists
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:١٥ م, Faraj Khasib fkha...@iconnecths.com wrote:
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor
/a/full into
/v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk
I attached log, but there is nothing unusual in it ...all normal ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com]
Sent: Wednesday, December 28, 2011 4:06 PM
To: Faraj Khasib
Subject: Your message to asterisk
Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib
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From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Monday, January 02, 2012 3:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call type in dial
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my
clients but the request is recieved at the other client as video call request
since I am enabling video support for sip
Sent from my iPhone
On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, Doug Lytle supp...@drdos.info wrote:
Faraj
and extensions.conf details which
is using for that communication.
And CLI output of asterisk is also required.
On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my
, virendra bhati
virbh...@gmail.commailto:virbh...@gmail.com wrote:
Which is means like if you are using sip 1234 then give the details of [1234]
into that open thread and relevent extensions details too
On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib
fkha...@iconnecths.commailto:fkha...@iconnecths.com
this is what my SIP Invite message when I make Video call
INVITE sip:6500@192.168.21.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
From: sip:6097@192.168.21.102;tag=1857098215
To: sip:6500@192.168.21.102
Contact:
in that call.
--
Regards,
Sammy
On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
this is what my SIP Invite message when I make Video call
INVITE sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933
Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
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] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call
Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Tuesday, 3 January 2012 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call
Actaully I didnt find a good
for example if I am using x-lite as client, how to I connect as guest from
client ...I am allowing guests at asterisk server
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha
anyone?
what should x-lite account be for guest user ?I tried guest but didnt work
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5
hey all,
My problem is that I am trying to have multiclients call my SIP queue, now each
client is not authorized so I tried to make them call using the same
extension but I got call overlap between all clients, now what I want is a way
that I can make all my client call the SIP queue
On 03-01-12 14:13, Faraj Khasib wrote:
anyone?
what should x-lite account be for guest user ?I tried guest but didnt work
A guest does not need an account on your asterisk server so you do not
need to configure an account on xlite. Instead on xlite you just dial
extension
at 4:01 PM, Faraj Khasib
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
thank you for your reply, but x-lite cannt dail without an active account
dail is disabled without any account
My problem is that I am trying to have multiclients call my SIP queue, now each
client
Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like
my command doesnt change anything
exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf
Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like
my command doesnt change
Providing which version of Asterisk you are using might be helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf
anyhelp guys?
I tried a lot of stuff
] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf
I tried also in asterisk 1.8 setting outbound variable but didnt work
also
https://wiki.asterisk.org
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf
how can u give me a command
Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf
didnt work also :(
From: asterisk-users-boun...@lists.digium.com
[asterisk
ON and see what codecs are available on
the other end.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Any suggestion will be great
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 11:55 AM
To: Asterisk Users Mailing List - Non
: Re: [asterisk-users] Set Call Codec in extension.conf
Please post the sip.conf entries for 6000 and 6500.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
allow=all
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf
Both sides?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:13 PM
To: Asterisk
is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
https://issues.asterisk.org/view.php?id=13243
Regards,
Sammy
On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib
fkha...@iconnecths.commailto:fkha...@iconnecths.com wrote:
thats excatly what I want
try the following
Set(${CALLERID}=722979797 722979797)
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com]
Sent: Thursday, January 19, 2012 8:03 AM
To: Asterisk Users Mailing List
or this
Set(${CALLERID(all)}=722979797 722979797)
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Thursday, January 19, 2012 8:47 AM
To: Asterisk Users
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is
called but it doesnt execute the command, I tried the command in terminal it
worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame -b 8 -t -F
-m
Hi guys,
I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk
1.6
but I cannt make it work, can any body help me plz?
Thank you
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