send me a copy of your sip config also
make sure dissallow is before allow.
Kind Regards
Andrew Colin
Technical Director
T:010 591 4358
C: 082 310 3007
but it seems that value of variable defined in external file is not getting
populated during the dialplan execution.
My example:
extract from one external file in /etc/asterisk/abc.conf
PROV=1.2.3.4
[abc]
exten = _1X.,1,Dial(SIP/${PROV}/${EXTEN})
and extensions.conf contains:
[globals]
hi Andrew,
here is my sip.conf
[]
host=x.x.x.x
qualify=yes
type=peer
insecure=port,invite
context=from-internal
disallow=all
allow=all
On Mon, Jul 29, 2013 at 9:17 AM, Andrew Colin and...@vsave.co.za wrote:
send me a copy of your sip config also make sure dissallow is before
allow.
I have a problem transferring calls multiple times using DTMF sequences (#, *2).
The scenario is:
Transfereecalls Transferor 1
Transferor 1 transfers to Transferor 2
Transferor 2 transfers to Transfer Target
When Transferor 2 enters '#' or '*2', Asterisk no longer reacts and the
You can take the pcap trace using tshark or tcpdump command line linux
based tool and open the trace in wireshark. Wireshak is visual tool of
tcpdum/tshark(corss platform) and you can listen audio of each call.
On Fri, Jul 26, 2013 at 10:17 PM, Gianluca Merlo
gianluca.me...@gmail.comwrote:
I just got access to an older Asterisk 1.6.2.18 box and found that the multiple transfer problem
does not exist here. So with 1.6.2.18 I can transfer as often as I wish using DTMF sequences.
jg
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-- Bandwidth and Colocation
remove disallow completely
you are basically saying do not allow anything
then allow anything
so remove the disallow part and leave allow
Kind Regards
Andrew Colin
Hi folks,
Recently a customer of us moved his old asterisk installation, an 1.4.44
to a VMWARE infraestructure and has started having some weird issues.
Asterisk started going slow and even refused to start up. After few tests,
it only loaded when deactivating queues and iax2 (with noload in
On 29/07/13 12:15, samuel wrote:
Hi folks,
Recently a customer of us moved his old asterisk installation, an
1.4.44 to a VMWARE infraestructure and has started having some weird
issues.
Asterisk started going slow and even refused to start up. After few
tests, it only loaded when
What is the output of g729 show version?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of james jan
Sent: Sunday, July 28, 2013 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello, working in a call center where we set up a structure in asterisk.
When my voip reaches 150 calls are with bad quality. We do not transcode
codec. What I realized using the top command server (CentOS) processing is
too high for the asterisk. But the general processor server is down. Would
Well, I forgot to add the t or T option to the dial command, which is required to do transfers
with DTMF sequences.
jg
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New to Asterisk? Join us for a
Dear asterisk users
I wanted to use E1 pri lines on my asterisk box but my provider support
only 120ohm on E1 line. I dont know how to set those values.
Please help me
--
Akib Sayyed
Matrix-Shell
akibsay...@gmail.com
akibsay...@matrixshell.com
Mob:- +91-966-514-2243
--
On 29/07/13 15:22, Eduardo Leones wrote:
Hello, working in a call center where we set up a structure in
asterisk. When my voip reaches 150 calls are with bad quality. We do
not transcode codec. What I realized using the top command server
(CentOS) processing is too high for the asterisk. But
On 29/07/13 16:28, Akib Sayyed wrote:
Dear asterisk users
I wanted to use E1 pri lines on my asterisk box but my provider
support only 120ohm on E1 line. I dont know how to set those values.
Please help me
Its done on whatever interface cards you have. Some may have a jumper
setting. I
Hi,
I've searched the asterisk.org and voip-info wiki sites, but not found an
answer that seems to match.
Hopefully this is a simple question. COLP is working very well on our
system - Unfortunately it is working a bit TOO well in some circumstances.
We have some untrusted trunks. On these
From: Steve Davies davies...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 07/29/2013 10:53 AM
Subject:[asterisk-users] Connected Line presentation in 1.8.x
upwards
Sent by:
On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 29/07/13 16:28, Akib Sayyed wrote:
Dear asterisk users
I wanted to use E1 pri lines on my asterisk box but my provider support
only 120ohm on E1 line. I dont know how to set those values.
Please
there's no dahdi installed.
Following debugging the issue, it looks like the astdb file is broken.
Whenever database show command is executed it loops over the same results.
The file itself is around 225K but dumping its content via asterisk -rx
'database show' creates and endless file.
Is there
hello:you can add T1_E1 by load card drivers
Best regards,
James.zhu
website: www.hiastar.com
From: akibsay...@gmail.com
Date: Mon, 29 Jul 2013 21:48:19 +0530
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] using E1 PRI lines
On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades
I didnt understand what you were saying.can you please explain
I am using digium cards
sent from android
On Jul 30, 2013 6:55 AM, James zhu zhulizh...@live.com wrote:
hello:
you can add T1_E1 by load card drivers
Best regards,
James.zhu
website: www.hiastar.com
On 30/07/2013, at 4:22 PM, Akib Sayyed akibsay...@gmail.com wrote:
I didnt understand what you were saying.can you please explain
I am using digium cards
sent from android
E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC
connectors ) or a 120 ohm balanced
Operators are unnecessarily confusing you by talking tech Lang which you
are not well versed with. Are you trying to create prod / services which
they don't want u to launch but they have to provide lines under some sort
of regulatory obligations ?
Just go ahead n plug the wires on the E1 card
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