yeah -- searching how to perform this magic ...
On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote:
On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
load balance incoming
:
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
load balance incoming calls over IAX2 trunks. If any trunk goes down the
calls traffic will be shared with other available trunks. When it gets Up
the script is supposed to perform as desired i.e in load balance mode.
you running GSM FWTs with asterisk ?
On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote:
HI,
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local
telecom provider for
Hi,
I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call hits
the box, the gets answered even the other end phone in not picked. How can I
fix this as ideally it should answer the call when other end phone is
picked.
--
Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the
box , it answers the call even the phone is not picked. ideally it should
answer the call when the phone is picked up. Its charging the clients.
Please let me know how can I cover this ? Thanks in advance.
--
Hi Olivier,
General Availability for snom8xx sidecar: ~March 2010
UT
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von
asterisk-users-requ...@lists.digium.com
Gesendet: 19 October 2009 15:15
An:
Hi Raimund,
snom uses basically the same concept. As explained under:
http://wiki.snom.com/Settings/user_failover_identity.
You select the line id that should be used when a registration fails.
Regards,
Usman
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
Unfortunately that is true for the time being. Since we moved our main
office to new premises, our telecom provider has failed setup services
in time. Forums and otrs is online and we hope to have the phones
working ASAP.
We appreciate your understanding.
Regards,
Usman
mechanism.
Regards,
Usman.
-
Usman Tahir
snom technology AG
www.snom.com
This e-mail may contain confidential and/or privileged information. If you are
not the intended recipient (or have received this e-mail in error) please
Hi Ron,
You can change this setting through the web interface Advanced/Audio/Dialtone
during Hold.
Hope that helps!
Regards,
Usman.
-
Usman Tahir
snom technology AG
Gradestraße 46
www.snom.com
This e-mail may contain
Hi Domenico,
Try Ver. 6.2.1. This problem is fixed in it.
http://www.snom.com/wiki/index.php/Beta_Firmware#Release_6.2.1
Regards,
Usman Tahir
snom technology AG
--
Message: 17
Date: Tue, 20 Jun 2006 18:18:43 +0200
From: Mimmus [EMAIL PROTECTED]
Subject
Detailed info about snom beta firmware can also be found at snom-wiki
e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes
Regards,
-
Usman Tahir
snom technology AG
and 180 as 180 Ringing
o WEB: enhanced french translation
-
Usman Tahir
snom technology AG
--
Message: 13
Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST)
From: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
Hi Brian,
For the conf on Xfer issue, use the latest beta
http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin
Regards,
-
Usman Tahir
snom technology AG
Gradestraße 46
D-12347 Berlin.
Tel: +49 30 398330
Fax: +49 30
Old Ringer 2 4 will be available as 9 10 (in addition to the
existing melodies) in Version 5.1 to be released in a few days. Its
better than wasting bandwidth downloading such a custom melody, as
Ringer2 seems so popular. Hope that will suffice...
Regards,
Usman.
Message: 13
Date: Tue, 3 Jan
bits (word) per sample.
Regards,
Usman.
-
Usman Tahir
snom technology AG
Gradestraße 46
D-12347 Berlin.
Tel: +49 30 398330
Fax: +49 30 39833111
[EMAIL PROTECTED]
www.snom.com
This e-mail may contain confidential
Title: snom Firmware 5.0.
Hi,
Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page.
Regards,
-
Usman Tahir
snom technology AG
www.snom.com
HI!
how are you people. i am a newbie in asterisk and
voip.
i need your help.
the scenerio is like this.
1.all local SIP users will be connected to asterisk
via IP.
2.PSTN will be connected to AS5300.pstn will give us a
local prefix like 333. so any one calling at
333 will go to my
Hi,
I need to run sip on non-standard port e,g 8881 and do not
want user to define this port in clients like ata or softphone.
what I want, when a client sends a register request at sip
server, the sip server should send him the port number OR is there a way
using DNS SRV
can any 1
anyone running SS7 with Asterisk ? Please help me out.
I need to know the hardware used for SS7 with Digium E1 cards...
Thanks,
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Is there any digium card that support E1 with SS7 and does Asterisk
support SS7 ???
any 1 who has done this ?
Usman
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Is there any digium card that support E1 with SS7 and does Asterisk
support SS7 ???
any 1 who has done this ?
Usman
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Hi !
When u enable queue monitoring application from queues.conf then u have to
specify a variable named MONITOR_FILENAME in extesnions.conf just before u
put the incoming call into the queue. This variable will contain the path
of the filename or the filename itself as with which u want to
Hi All,
I am having trouble with MOH. I have downloaded the latest CVS head and
when I try to call from PSTN side and play MOH on a queue then the voice
breaks. However if I play the same file using Playback() application and
listen to it through PSTN side then there is no problem. CVan
Hi Wiley,
thanks for ur reply ! yes ! I am using custom music files. And they are
not mp3 rather they are .wav files. Even if I use mp3 files the problem
remains the same. And about transferring them I use SCP to transfer files
which internally uses SSH i guess. I am not sure about MOH volume.
Hi
I have applied for Qwest 1800 termination to a T1 ISDN PRI. They tell me
that I will have to program a predefined DNIS number on my switch.
According to them unless asterisk returns that DNIS number no call will
get through.
How do I program the DNIS, is it through zaptel.conf or some other
Hi All!
I am using Asterisk Stable 1.0.6 . Now I want to add another application
like app_chanspy in it. I have downloaded its source file but how can I
merge this application along with my already running asterisk ? Any
comments suggestions are appreciated ...
Thankyou,
Usman
Operator Panel but it
works only if two asterisk SIP extensions are calling eachother. It
doesnot work in the case if one of the call comes within from a
queue. Any tweaking in extesnions.conf that could help me figure this
out Any useful help , comments are appreciated ... thanks.
Usman
hangs up
without recording anyhting. And if I put the Monitor application on top of
Queue command then I have to specify the saving filename before I know
that to which agent the call is going. ANy comments , suggestions
appreciated.
Thanks,
Usman
hi all,
I have got a problem with asterisk fromuser field in sip.conf. Actually
I have got two asterisk servers communicating over sip. When a user from
Asterisk Server A calls a specific extension it is redirected to another
Asterisk Server B and that Asterisk Server B forwards it to a
= 8000,3,Queue(supportq|t)
plz help me inthis regard ... Thanks !
Usman.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
might
be wrong ?
thanks !
usman.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
functionalities.
Can anybody guide me how to make attended call transfers work in this
scenario if the SIP phone doesnot support attended call transfers. I'll be
waiting for any valueable feedback.
Thanks,
Usman.
___
Asterisk-Users mailing list
[EMAIL
]:5061,tr
exten = _1.,2,Hangup
Please help me in this reagard.
Regards ,
Usman.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
to be done in extensions.conf to make it
work ? plz help me in this regard.
Usman.
This patch works a treat for us:
http://bugs.digium.com/bug_view_page.php?bug_id=0002460
Makes all # transfers attended, but the transfer button on the phones
can still be used for blind transfers from our
36 matches
Mail list logo
