Hi,
I am having trouble passing variables via the call files, here is my call
file via the php:
fputs($oSocket, "Action: login\r\n");
fputs($oSocket, "Events: off\r\n");
fputs($oSocket, "Username: $strUser\r\n");
fputs($oSocket, "Secret: $strSecret\r\n\r\n");
fputs($oSocket, "
Is it possible to do SIP<->Asterisk<->TDM in a single step with FFA? Or
does FFA always use TIFF files?
I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102
ATA's at the fax machines and send faxes directly over a PRI.
Mark
--
Mark Willis
Star One Telecom
Office: 1-800-889-70
${BLINDTRANSFER} should hold the device name of the one doing the
blind transfer.
On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock wrote:
> Hello!
>
> I am trying to find out the device name and/or other identifying data
> to be used in a context when a device transfers the call to new a
> phone
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote:
>This works for me.! but the agent has to dial the number ?
>How could be the context for do this ? U can give an example ?
I'm using this to place calls from local IP-phones over the PSTN. So my
script will generate, say:
Channel: SI
My problem is that I dont know how to do for transfer the call to agentExample,
I have this .call
Channel: Zap/g1/8652323454MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context:
call-file-test Extension: 10
So my context is this
[call-file-test ]exten => 10,1,Dial(SIP/2031,tT)exten => 10,2,hangup
I
Hello!
I am trying to find out the device name and/or other identifying data
to be used in a context when a device transfers the call to new a
phone number. From running tests, it looks like the account code
variable (${CDR(accountcode)}) is set to the account code of the
device that placed the o
> Date: Sat, 12 Feb 2011 21:35:29 +
> From: ro...@firedrake.org
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Using files .call or AMI
>
> On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote:
> >I have a webpage with information about a customer so in thi
On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote:
>I have a webpage with information about a customer so in this page the agent
>click a phone number and asterisk do the call and transfer the call to agent
>if this call is answered.
Usually it's the other way round: the agent's phon
as you know you have 2 ways. using ami or .call files. if you
have experience, the AMI is more powerful.
you must have a context in your extensions.conf to manage agent procedures,
it looks like a simple context, that you must have, for managing queues.
with .call file or ami dial your customers,
Hi!
I have a script to generate calls from a database using .call files and giving
a message. If works great! but now I need to do the same but instead of play a
recorded message I need transfer this call to live person in a specfic
extension. This is the scenarioI have a webpage with informati
On 02/11/2011 07:56 AM, Albert wrote:
can anymore drop me a asterisl's config for digium te220b (with ec) or
at least some good tutorial of configuratin e1 line with that card ?
The information you are looking for (for Asterisk, not for 'asterisl')
is provided in the manual for the card; if y
On Sat, Feb 12, 2011 at 10:11 AM, Terry Brummell wrote:
> Yes, I use provisioning for my Polycom's. And unfortunately, as far as I
> know, the Mitel's do not support tftp/http provisioning. I did just upgrade
> my 5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I
> don't
Thanks for the comments, I will go through the detail and price and then
will buy accordingly,
cheers
/ag
On Sat, Feb 12, 2011 at 2:11 PM, Terry Brummell wrote:
> Yes, I use provisioning for my Polycom's. And unfortunately, as far as I
> know, the Mitel's do not support tftp/http provisioning.
Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know,
the Mitel's do not support tftp/http provisioning. I did just upgrade my
5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't
know what the phone is trying to do in that folder.
Anyway, that
On Sat, Feb 12, 2011 at 9:50 AM, Terry Brummell wrote:
> Aastra & Polycom because they can be configured using a TFTP server. Great
> for large installations with centralized management.
>
>
>
> Mitel 5215/5224 because they are dead simple to configure (via web gui) and
> just plain work with no
Aastra & Polycom because they can be configured using a TFTP server.
Great for large installations with centralized management.
Mitel 5215/5224 because they are dead simple to configure (via web gui)
and just plain work with no screwing around.
From: asterisk-users-boun...@lists.digium.c
On Sat, Feb 12, 2011 at 9:31 AM, ast guy wrote:
> Hi,
> I have been out of touch with asterisk for quit some time and needed some
> recommendations. I am looking for SIP hardphone that works well with
> asterisk server.
>
> Pls suggest.
>
> cheers
> /ag
Currently most every phone works well, if
zyxel
On Sat, Feb 12, 2011 at 4:01 PM, ast guy wrote:
> Hi,
> I have been out of touch with asterisk for quit some time and needed some
> recommendations. I am looking for SIP hardphone that works well with
> asterisk server.
>
> Pls suggest.
>
> cheers
> /ag
>
> --
> __
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Pls suggest.
cheers
/ag
--
_
-- Bandwidth and Colocation Provide
Apologies, using two underscores (I retested) did not cause the error
On Sat, Feb 12, 2011 at 1:42 AM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:
> Alrighty Gents, let's see if any of you have encountered this
> one...Variables losing their value...I'm setting a variable with four
> un
On Sat, 12 Feb 2011 10:14:42 +0100, Gilles
wrote:
>Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends
>FXO calls to a context named "numberplan-local" that is not mentionned
>in my configuration file, which prevents incoming calls to be
>successfull:
Found what it was: This co
Hello
Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends
FXO calls to a context named "numberplan-local" that is not mentionned
in my configuration file, which prevents incoming calls to be
successfull:
=== /etc/asterisk/zapata.conf ==
[trunkgroups]
[chann
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