On Wed, May 26, 2010 at 04:41:57PM +0100, salaheddine elharit wrote:
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
What's dialplan.conf ?
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers
Hello everyone,
any help please
I have asterisk installed in our call centre with aheeva platform and
centos linux,
We have 2 access provider I have configured the
etc/asterisk/extensions.conf in order to do the routing of calls
exten = _0612.,1,Set(CALLERID(number)=520460587)
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1
instead g2
Any help please will be appreciated
Maybe if you asked a question, something could help. But, as it is
: Wednesday, May 26, 2010 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] routing of calls
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed
Danny Nicholas wrote:
Doug, did you cancel your psychic friend's subscription? All programmers
are supposed to be able to determine intent without full information :)
I had too! I'm on a budget and it was costing me more then my cable bill.
Doug
--
Ben Franklin quote:
Those who
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers take the secend
provider
; Outbound dial context
[aheeva_ccs]
; If we are dialing out through
I dont know, maybe I am missing it. I see nothing off the top of my head that
shows you attempting to dial out 2 different providers or fail between them.
Both times you have posted code I see a dial command set to go to a single Zap
Group, and no failure code or Prefix that determines how or
Hello everyone,
I have asterisk installed in our call centre with aheeva platform and centos
linux,
We have 2 access provider I have configured the etc/asterisk/extensions.conf
in order to do the routing of calls
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten =
:56 AMSubject: Re: [Asterisk-Users] Routing SIP calls via URIBut is there a way of doing this without a prefix?because people should dial without prefixes: "[EMAIL PROTECTED]" , not like:"[EMAIL PROTECTED]"How can we make this without a prefix? something
like:if( !uri=~"
@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten =
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk
Dear All,
I have the following setup;
SER/External Asterisk -- Firwall -- Internal Asterisk -VPN- Users
At the moment;
Anybody can register with our SER proxy and call each other using VoIP.
Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our
Shad Mortazavi wrote:
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk
Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Date: Wed, 29 Mar 2006 13:18:07 -0600
Shad Mortazavi wrote:
What I would like to do is to redirect external
On Mon, 17 Oct 2005 00:05:39 -0400
Tom Rymes [EMAIL PROTECTED] wrote:
Your other options include FXO gateways like the sipura 3000 9which
is an ATA, too), Digium TDM400p PCI card, or a T1 card and a channel
bank.
The appropriate piece of equipment depends on the number of lines you
Is there possible to route ordinary landline-calls to the asterisk server
and from there too our SIP-phones using a regular 56000 bps modem?
--
MVH
Peter Ankerstål.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing
Short answer: No, but... Long answer: Yes, and...
Essentially, there are *certain* internal modems that will handle this function,
but basically what you're talking about is an FXO card. You can pick up one for
little outlay on eBay.
Do a search on eBay for X100P. Then read the wiki for
Your other options include FXO gateways like the sipura 3000 9which
is an ATA, too), Digium TDM400p PCI card, or a T1 card and a channel
bank.
The appropriate piece of equipment depends on the number of lines you
will need.
Tom
On Oct 16, 2005, at 6:49 PM, Rod Bacon wrote:
Short
-Commercial Discussion
Subject: RE: [Asterisk-Users] Routing DID calls to external lines
Try answering the line first.
Exten = 500,1,Answer()
exten = 500,2,Dial,Zap/g1/3105551010
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Syed Akbar
Sent: Friday
I am trying to route incoming DID call (on a analog channel) through
Asterisk to an outside (analog) line. My extensions.conf is something like
the following:
exten = 500,1,Dial,Zap/g1/3105551010
In this case the incoming DID call extension is 500. I am able to dial out
and connect with
Discussion'
Subject: [Asterisk-Users] Routing DID calls to external lines
I am trying to route incoming DID call (on a analog channel)
through Asterisk to an outside (analog) line. My
extensions.conf is something like the following:
exten = 500,1,Dial,Zap/g1/3105551010
In this case the incoming
I am trying to route incoming DID call (on a analog channel) through
Asterisk to an outside (analog) line. My extensions.conf is something like
the following:
exten = 500,1,Dial,Zap/g1/3105551010
In this case the incoming DID call extension is 500. I am able to dial out
and connect with the
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
Also, is it possible to put a phone into multiple contexts?
For instance:
of the last three known GPS coords (cell sites)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Saturday, March 19, 2005 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
Matt wrote:
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Also, is it possible to put a phone
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing
done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Be careful. 911 centers are not
I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality. Can someone point me in the right direction?
Script Head
Do a search on ACD and agents, this is certainly achievable.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis
Voitenko
Sent: Friday, January 14, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Routing
Subject: [Asterisk-Users] Routing incoming calls to various extensions.
I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality
I'm new to this list.Reading the asterisk
handbook pdf (good work)but but still have some questions.
Using Trustix 2.1 and installed Asterisk via CVS,
zaptel and libpri.
We have a dedicated server which is connected to
our telephone company.
It makes us able to call ordinary phones via VOIP
Thune
Sent: Monday, August 23, 2004 6:39 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] routing telephone calls via
switchboard/asterisk.
I'm new to this list.
Reading the asterisk handbook pdf (good work) but but still have some
questions.
Using Trustix 2.1 and installed Asterisk via CVS, zaptel
: [Asterisk-Users] routing telephone calls
viaswitchboard/asterisk.
Yes, it's very likely that you can perform these IVR functions within
asterisk.
If the realtime switching decisions are simple, they can probably be
stored in the asterisk dialplan itself. Alternatively, you could retrieve
them
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