get a nasty rate spike?Abid -> If we go over,
the call will not be terminated nor the next origination will fail nor we will
get nasty rate spike but they will send us warnings and then finally block our
trunks.
Best
Regards,
---
Abid
Saleem
Technical Manager
Dear AJS,
Thank you for your response with good idea. Unfortunately I am not good at
programming. Can you please write this AGI script for me. Please help if you
can.
Best
Regards,
---
Abid
Saleem
Technical Manager NGN
Terminus Technologies
Mobile: +92
Hi All,
I have 100 Trunks from my Provider. My Provider is restricting me to make only
120 minutes Call duration / trunk / day. So I want to load balance my calls to
these 100 trunks. Please advise in this regard ASAP. Thanks in advance.
Regards---Abid SaleemTechnical Manager NGNTerminus
with that trunks User name
in Preferred-Identity header.
Regards---Abid Saleem
> Date: Wed, 18 May 2011 10:07:03 -0400
> From: abalas...@evaristesys.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Failover trunks
>
> On 05/18/2011 10:05 AM,
I configure extensions.conf
for calls to be sent to the next available trunk.
Please help.
Regards-Abid Saleem --
_
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Hi,
I am using Asterisk 1.4.17 for my C4 routing but I am experiencing a high pdd
of around 20 seconds. Could you please help me to reduce it and what could be
the reason. Thanks
Abid Saleem
Hi,
Please help me with this.
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. My call comes from my wholesale client and lands on my switch, then
it is routed to asterisk. I want ast
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 termination. MY call comes from my wholesale client and lands on a
switch, then it is routed to asterisk. I want asterisk to ro
have a look on feature.conf and implement feature in [applicationmap]
dialfeature => ##,peer,DIAL,{DATA_OF_DIAL_NUMBER_WITH_TECHNOLOGY}
regards
Dhaval
On Thu, Mar 31, 2011 at 7:09 PM, Abid Saleem wrote:
Hi,
Does anyone know how to implement the feature in asterisk calling card when a
u
Hi,
Does anyone know how to implement the feature in asterisk calling card when a
user has dialed the access number and during the IVR or any time during the
call, he can press ## or ** to end the current call and dial a new destination
number?
Please help and give me a step by step help. Thank
minute of the call.
Please someone help how to achieve this. Thanks.
Regards-Abid Saleem
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
--+But
it still has the same problem. Any further suggessions. Thanks.
Abid Saleem
> From: [EMAIL PROTECTED]> To
May be. If somebody has experience this problem before, then only he/she can
guide about this. I am not sure whats going on.
Abid Saleem> Date: Mon, 4 Aug 2008 09:19:45 -0400> From: [EMAIL PROTECTED]> To:
asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Asterisk
R
MySql only.
Please help me. Thanks in Advanced.
Abid Saleem
Product Engineer VoIP Solutions
Comcerto Bahrain W.L.L
_
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBR
configuraton.
Please help.
Thanks.
Abid Saleem
_
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___
-- Bandwidth and Colocation
Server. Please suggest urgently.
Thanks.
Best Regards,
-
Abid Saleem Choudhary
Team Lead VoIP Networks
Comcerto Bahrain W.L.L.
Direct: (973) 13301504
Mobile: (973) 36080504
Tel: (973) 13301100, Fax: (973) 13301101
MSN: [EMAIL PROTECTED]
WebSite: http
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