Hi All,
In my dialplan and standard asterisk CLI logging i see that i am able to
restrict the callerid when dialing out with asterisk.
however, on the receiving phone, the callerid is still displayed.
When i increment the logging of the pri with pri set debug on span 1 on the
CLI i also get
Any particular reason you don't want to put the logic of the macro in your
AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking
how to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias
I have a pair of Asterisk servers which are happily routeing VoIP calls.
I want to hook one of them to the PSTN. Given that I am in the UK, what
is a reasonably easily-available device to provide an FXO interface from
a Linux box, with a minimum of faffing around with drivers? Just one
line is
On 3 Sep 2010, at 10:07, Roger Burton West wrote:
Also: I've heard good things about the PAP2T for getting analogue
handsets to talk to a VoIP server. But all the ones I can see on eBay
are PAP2T-NA models. Will these work with British handsets? (Obviously
with a plug adaptor to put the BT
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
Subject: [asterisk-users] openvz
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
Blind Answer - you should be able to; Asterisk doesn't rebuild the
I use Asterisk 1.6.2.11 and this is my dialplan:
[test]
exten = ,1,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,Answer()
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,PlayBack(hello-world)
exten =
On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote:
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
I don't expect any problem.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.4.0.
DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are
available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.
I am having trouble completing faxes. Carrier send calls to me using SIP.
Any recommendation to have some success with Fax.
We trying using T.38 pass through and using G711U codec.
Asterisk Version 1.6.1.1
On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote:
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
I don't expect any problem.
Absolutely right: 1.6.x works fine with OpenVZ and Virtuozzo out of the box
as long as you don't need any hardware interfaces.
I've done it ;)
This is what i did:
In the Macro:
[macro-check-call-limit-mercurios]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}]?forbidden,1)
; EXITO:
On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com wrote:
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.
I am having trouble completing faxes. Carrier send calls to me using SIP.
Any recommendation to have some success with Fax.
We trying
Roger Burton West wrote:
I want to hook one of them to the PSTN. Given that I am in
the UK, what is a reasonably easily-available device to
provide an FXO interface from a Linux box, with a minimum of
faffing around with drivers? Just one line is needed, though
in theory two might
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
The carrier sending the calls wants me to be able to pass faxes to physical
fax machines on the PSTN. So far they are failing.
We just want
Outlook?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Friday, September 03, 2010 3:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users]
You can come and take my 3 Ayaya switches and all the associated cost$!
It'a all * for me.
and to answer your ?---they cost too much$
=)
On Thu, Sep 2, 2010 at 12:32 PM, bruce bruce bruceb...@gmail.com wrote:
I am not interested in open source solutions. I want to know how much the
1- I am interested in this as well. Looking into Proxmox as it provides a
nice interface (do you guys know of any other good one?)
2- Would the conference calls be fine as well? I understanding Asterisk
1.6.x uses a kernel timing source now a days so that ztdummy is not needed
anymore?
3- Would
Blind Answer - you should be able to; Asterisk doesn't rebuild the
kernel. You might have to get some kernel source using ZYPPER (in caps
so Outlook express doesn't change it to zipper).
El 03/09/10 09:31, mattias escribió:
Outlook?
Outlook Express is a total PITA. Should I
On Thu, Sep 2, 2010 at 2:26 PM, Thorolf Godawa nos...@godawa.de wrote:
Any idea what is going wrong here?
Read doc/backtrace.txt
If you cannot get Asterisk to coredump, try running it under gdb to
see what is happening.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
You don't say the percentage that are failing. However, people
All my attempts are failing.
Thanks
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, September 03, 2010 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Can you post the dialplan snippet you are using?
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
On 09/03/2010 10:50 AM, dave george wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
The carrier sending the calls wants me to be able to pass faxes to physical
fax machines on
Thanks Kevin,
I tried passing it over VOIP using g711U codecs with no success. I will try
using the patches that you mentioned and post the results.
Thanks,
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext
g711 across a network without perfect jitter/delay characteristics will not
work.
You cannot do g711 faxing across the internet - at all.
It's not a perfect solution even in an office on a dedicated LAN environment
(you'll still get failed faxes).
On Fri, Sep 3, 2010 at 12:32 PM, dave george
Some days ago in my lab I setup Proxmox, installed a CentOS 5.2 appliance on
OpenVZ, installed all asterisk related stuff (except dahdi), including php,
mysql, munin, other tools, set it up with a dialplan and it worked just
fine. Then manually made multiple copies of the folder where all this
Try open souce solution ICTFAX for T.38 faxing developed by us available
at http://www.sourceforge.net/projects/ictfax
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak jmas...@antelope.net wrote:
g711 across a network without perfect
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