Displaying 20 results from an estimated 5000 matches similar to: "Call files"
2007 Oct 31
4
AEL2 and Callbacks
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 60000
Exten: callback
Channel: Local/6505551212 at LegA
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
-----------------------
context LegA {
_X. => {
Dial(SIP/${EXTEN}@Provider);
}
}
And my default context:
2016 Oct 17
2
Streaming for ASR
Matt Riddell wrote:
>
>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com
>> <mailto:luca.pradovera at gmail.com>> wrote:
>>
>> I have been working on designs for two different projects, where both
>> of them would need to use the IBM Watson streaming ASR service.
>>
>> Would it be possible to write out the audio frames
2007 Aug 15
2
Load balancing SIP trunks?
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there any easy ways for Asterisk to either balance between
SIP trunks or even just a built in function to find the next available SIP
trunk. I think using
2017 Feb 06
3
Call List Campaign to an IVR
> On Mon, 6 Feb 2017, Tech Support wrote:
>
> We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2)
> delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were
2007 Apr 16
3
Redundant * servers
Without using Dundi or SER, any thoughts on the following anyone?
Has something similar been implemented anywhere so as to me not
having to horribly butcher code...
4 servers SIP1-4
User1 -- -- SIP1 --
\ / \
User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB
/ \ /
User3 --
2017 Feb 06
3
Call List Campaign to an IVR
Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg.
If you wanted you could leave it ringing for twenty minutes and it would still have the same effect.
Kind regards,
Matt
> On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote:
>
2010 Aug 15
6
Realtime Context
Hi,
I'd like to be able to create contexts in real-time when I add new clients to my asterisk box.
Currently, I have to create a blank context in extensions.conf and add:-
switch => Realtime/@
Is there any way to avoid the step of creating the blank context and simply include all the entries from the database?
Thanks
Dan
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2004 Aug 17
1
Samba3 and eDirectory as LDAP - HELP
I've got problem with Samba3 when I want to connect to eDirectory. The samba
could not connect to the local eDirectory server. The Samba and the eDir can
be found on the same Linux box which is a SusSe 9.1. The eDir and the Samba
are working pretty good but cant see each other. Is it possible to connect
from Samba 3 to Novell's eDirectory ? What are the syntaxes to these lines
in smb.conf:
2007 Apr 15
9
Loudspeaker
Hello List,
This is what I want to do:
When a call comes in I want to ring an extension that happens to be loud
speaker. The users can the press *8 to answer the call. Is there a
SIP device that I can connect to Asterisk as an extension that can
accomplish something like this?
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2009 Jan 12
6
CDR Rewrite -- Questions to the users
Hello!
Most are probably bored seeing another letter about this,
but I've put in a fair amount work on a spec for rewriting
the CDR system in Asterisk, and I have some questions:
First, please look at what I've written so far:
svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs
and look at the file "CDRfix2.rfc.txt" in the RFCs dir.
The spec SIGNIFICANTLY alters the way
2007 Jun 27
1
Round Robin SIP peers?
Hi all,
I have a cheapskate customer whom wants to leverage some cheap
all-you-can-eat VoIP connections rather than pay for a per minute
provider.
On the inbound side I think I have a solution in that I can activate the
"call forward on busy" option with his provider (some noname white label
house) but how do I balance his outgoing minutes?
Is there some way that I can set up a round
2005 Nov 24
2
Fwd: Matrix rotation
Ok I warned you that I'd been drinking! What I really meant was
something to go from:
[,1] [,2]
[1,] 1 2
[2,] 4 3
to
[,1] [,2]
[1,] 4 1
[2,] 3 2
to
[,1] [,2]
[1,] 3 4
[2,] 2 1
to
[,1] [,2]
[1,] 2 3
[2,] 1 4
Sorry for being a muppet, B
Begin forwarded message:
> From: Benjamin Lloyd-Hughes
2007 Mar 27
3
ztdummy and MOH
Hi All,
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no
Digium cards. The problem I have is that MOH will not play. It starts
and then stops.
asterisk*CLI> zap show status
Description Alarms IRQ bpviol
CRC4
ZTDUMMY/1 1 UNCONFIGUR 0 0
0
I'm not sure if the above is correct.
2007 Aug 01
3
TE120P in Canada
Hi All,
I'm having problems trying to get a TE120P operational in Canada.
I keep getting a congestion error when I try to make a call. I'm not
sure if my switching, parity, etc is correct. I'm hoping that someone
will be able to verify my config.
The Telco is SaskTel, with a 10 channel 50 DDI service.
Zap show channels show and ztcfg -vv looks ok and the zttool show
2008 Oct 07
1
can't find mysqlclient : asterisk-addons-1.6.0
Hi All,
I can not install the asterisk-addons as it thinks there is no
mysqlclient installed. I have installed mysql, mysql-server and
mysql-devel and I am still unable to install the addons. I am running
CentOS 5.2 i386.
Please somebody help.
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2013 Feb 19
2
Call Pickup how to display CND of incoming number
Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset?
I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call.
Regards
David
2003 May 13
2
Nortel i2004 IP Phones
Hello,
is it possible to connect a Nortel i2004, (i2002) IP Phone to an *
PBX?
I have seen on the net that there should be an SIP firmware aviable for
the phone, anyone an idea where to get it? Because orginally they use a
own protocol
to connect to an Meridian, BCM or CSE1K.
Another idea is to write a module for * that can handle this protocol.
bye
SLiME
2006 Dec 10
4
X100P clone dial problems.
I'm not sure if I have a configuration problem or not. I am unable to
dial out. When I try to dial in I can hear the phone ring on the
dialling phone but Asterisk does not register anything.
In zaptel.conf I have
loadzone = au
defaultzone=au
fxsks=1
In zapata.conf
language=au
context=from-pstn
When I do: zap show channels I get:
Chan Extension Context Language
2005 May 31
3
Use NTFS Partition.
Hi!! I have a CentOS 4.o install on some machine in my job. I need setting
the system for read/write partition NTFS. I know the kernel-2.6.x this
support is part of the kernel, but in CentOS this support is unaviable.
How can aviable this support whit out recompiler kernel?? Some rpm packege
for do that??
Regards,
David
_________________________________________
Tec. David Gonzalez Romero
2004 Apr 15
2
T1 Line install.. (UK Muppet)
Hi all, Muppet from the UK asking for help
We are just about to have a T1 line installed in our office in Dallas
and "Advantex" the supplier has sent a questionnaire asking a number of
questions. I have put the question area at the bottom of the email, we
will be using Digium's hardware. could anybody help :-)
In the UK when I asked for a E1, number of trunks required and the