Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ?
And how to integrate
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail:
Yes, but i tried to use Iaxmodem, but there is the mechanisms for the
CLASS 0 (data).
From: Dave Fullerton dfullertaster...@shorelinecontainer.com
Subject: Re: [asterisk-users] Asterisk and Software Data Modem
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi there
I am constantly running into wall, Asterisk version 1.6.1.6 is segfaulting
if I run more then 10 cps with 10sec call duration with sipp. For example
21cps rate segfaults on invite number 213, but if I use 10cps I have no
problem at all.
So i then tried with 1.6.2.0-rc3 and it can
Dear All,
In Perl AGI, I have two number like 700, 800. I have to call first 700.
Next I have to call 800. After that I have to connect this two numbers in
the call. How can I do it in Perl AGI?
Please anyone provide some idea...
Thanks,
Velusamy
Hello all,
I have a pretty much standard installation of an Asterisk 1.6.1.6 with no
PRI cards of any type (full VoIP).
Occasionally (it happens every 2 weeks or so), it just stops running. I was
using safe_asterisk but it seems that safe_asterisk did not restart it. I do
have the core dump file
I've a strange problem with CallerID when calling Vodafone mobile's from
my Asterisk Box.
If I dial out on my ISDN-30, setting the CallerID to my DDI (XXX802), a
Vodafone mobile correctly shows the incoming call with this number and
if the phone's not answered it shows a missed call from XXX802.
Hello, I am having a problem with getting call transfer to work.
This is what is happening:-
1) External call comes in on SIP from a DDI provider
2) The call is answered by extension 204
3) Then extension 204 presses the Xfer button and the call is
placed on hold
4)
On Wed, 4 Nov 2009, velusamy velu wrote:
In Perl AGI, I have two number like 700, 800. I have to call first 700.
Next I have to call 800. After that I have to connect this two numbers
in the call. How can I do it in Perl AGI?
You can create a call file or use AMI (requests via TCP socket).
Right now, I think it does not.
Look out for it at: asterisk-...@lists.digium.com
Regards,
Juan
Khaled W Chehab wrote:
Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP)
protocol ?
And how to integrate
Regards
Khaled
Chehab
NGN Eng.
On Wednesday 04 November 2009 04:23:16 Josip Djuricic wrote:
I am constantly running into wall, Asterisk version 1.6.1.6 is segfaulting
if I run more then 10 cps with 10sec call duration with sipp. For example
21cps rate segfaults on invite number 213, but if I use 10cps I have no
problem at
Hello,
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.
These guys are pretty close. http://www.ss7box.com/
On Wed, Nov 4, 2009 at 4:16 AM, Khaled W Chehab kche...@xplorium.comwrote:
Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ?
And how to integrate
Regards
*Khaled Chehab*
* NGN Eng.*
That's what I thought. You are on system A and trying to call an extension
on system B? When you call in from the PSTN, you are in a bridged
environment and moh is playing as a function on the bridged channel. When
you do the internal call, no bridge exists. Can you post the CLI output
from
Hi,
we're using Asterisk 1.6.1.1.
We've got a problem during a transfer call concerning the destroying of
the unused channel.
At the beginning we have a call A - B (A and B being two user).
With the AMI interface we ask the redirection of:
- case 1: the caller (user A)
- case 2: the called (user
Hi,
we're using Asterisk 1.6.1.1.
We've got a memory leak with unconnected static users.
Our template for static users in users.conf is:
[static](!)
hassip=yes
hasiax=no
type=friend
[100](static)
context=periferico
host=192.168.0.1
port=5060
username=100
[101](static)
context=periferico
Hi Danny,
Sorry for the late reply. You understood it exactly right, but there
is a problem. I don't have agents, I have members. I would like
something similar to the application AgentCallbackLogin. I am not
using AgentCallbackLogin because it was removed in asterisk 1.6.
I changed my
velusamy velu schrieb:
In Perl AGI, I have two number like 700, 800. I have to call first 700.
Next I have to call 800. After that I have to connect this two numbers in
the call. How can I do it in Perl AGI?
I think you are looking for the Bridge manager command which is
available since
I've opened a few bugs on ExternalIVR and added patches.
The biggest issue is:
https://issues.asterisk.org/view.php?id=16174
[patch] ExternalIVR does not handle arguments in a consistant manner
Basically, this optimizes and fixes several different ways of calling
ExternalIVR. If there is anyone
On Wed, 2009-11-04 at 15:16 +, Alexandre Rodrigues wrote:
I changed my call-limit to one, and the same problem continues when I
restart asterisk. Have you any moore ideias to solve this problem?
There's a note in the sample queues.conf configuration file (at least in
the 1.6.2 branch)
Hi,
Finally I made the upgrade. Everything work well, but I have an issue with
one of the G729 licenses, I can't load it.
I have two other license files that load with no problems.
The host id is the same in all the files, can you give me an idea how to
check this problem?
Kevin P.
On 11/04/09 08:24, Danny Nicholas wrote:
That's what I thought. You are on system A and trying to call an extension
on system B? When you call in from the PSTN, you are in a bridged
environment and moh is playing as a function on the bridged channel. When
you do the internal call, no bridge
Hi,
If by chance you should find your self in Paris or wish to be there to
present... this is for you.
Note they do NOT want commerical presentations and this is only about
Open Source Asterisk
http://www.astrieurop.com/en/
I am considering going. Digium being a premier sponsor, I imagine some
Luis Silva wrote:
Finally I made the upgrade. Everything work well, but I have an issue
with one of the G729 licenses, I can’t load it…
I have two other license files that load with no problems.
The host id is the same in all the files, can you give me an idea how to
check this problem?
Hello,
Several folks working with Kamailio, SIP Router, SER, OpenIMSCore,
SEMS and Asterisk are in Berlin next week, so we think of having a
dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens
that you are around and want to join, please send me an email to make
sure you
Hello,
I've been tasked to look for ways to resell to others the service that
one of a trunk provides.. In other words, i want to configure my
current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act as a
trunk to others.. I would provide an IP to them from one of my servers
and
Asterisk Project Security Advisory - AST-2009-008
++
| Product| Asterisk|
Asterisk Project Security Advisory - AST-2009-009
++
| Product| Asterisk|
I'm looking to build a VoIP solution for 100+ service vehicles that have
WiFi hot spots installed (with cellular uplinks). Currently we are
trying out Skype wireless handselts and Majick Jack. I'd also like to
consider an Open Source solution that can bring the calls back to our
data center
The Asterisk Development Team has announced security releases for Asterisk as
the following versions:
* 1.2.36
* 1.4.26.3
* 1.6.0.17
* 1.6.1.9
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of 1.2.36 resolves
Have you considered putting an advertisement in the newspaper?
PaulH
On 05/11/09 06:43, Carlos Cuervo wrote:
Hello,
I've been tasked to look for ways to resell to others the service that
one of a trunk provides.. In other words, i want to configure my
current Asterisk (Ver. 1.4.26.1) with
On 11/04/09 15:20, Adam Tauno Williams wrote:
For hardware someone on the IRC channel suggested the Cisco SPA3102 and
a cordless phone? This devices appears to be able to register to an
Asterisk server (there are many related forum and BLOG posts). Does
anyone have an opinion for or against
Sorry Paul.. Am not sure i understand your response.
Carlos
On Nov 4, 2009, at 3:44 PM, Paul Hales wrote:
Have you considered putting an advertisement in the newspaper?
PaulH
On 05/11/09 06:43, Carlos Cuervo wrote:
Hello,
I've been tasked to look for ways to resell to others the
Tzafrir Cohen wrote:
On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote:
Hi,
Yesterday I've got a core dump from Asterisk, other times I was able to
discover what this core dump was related with through gdb Ouput info,,
but this time.. I'm really lost. Could some one please
Continuing the siren14 usage thread:
sip.conf has:
disallow=all ; First disallow all codecs
allow=siren14;
Should I be able to originate an outbound call with siren14 as my only
codec?
When I try originate using either the spool file or a CLI originate
On Wed, Nov 4, 2009 at 1:44 PM, Joseph syscon...@gmail.com wrote:
On 11/04/09 15:20, Adam Tauno Williams wrote:
I have two of these and experience a lot of echo on PSTN line (FXS line works
OK).
The echo is almost impossible to get rid of, so test it first before you buy
this unit; Google
What are you reaching out to exactly? It would need to be a Siren14
capable. Also, do you have the Siren codec binary installed? It's not
part of the Asterisk distribution.
Also, you should know that all Siren14 calls are presently downsampled
to 16 KHz, so are effectively Siren7.Asterisk doesn't
On 11/04/09 16:01, Andrew Hakman wrote:
On Wed, Nov 4, 2009 at 1:44 PM, Joseph syscon...@gmail.com wrote:
On 11/04/09 15:20, Adam Tauno Williams wrote:
I have two of these and experience a lot of echo on PSTN line (FXS line
works OK).
The echo is almost impossible to get rid of, so test it
Did you try changing the FXO port impedance? Did you try resetting the
unit to factory defaults (on the IVR dial 73738# and then 1 to
confirm)? Did you by it from ebay and it shipped from hong kong or
similar? Did you try the linksys silver one on the same line as the
real Sipura one was on?
I
I'm confused about the need for an ATA with an FXO port in a car?
Phone - PAP2T - ETH - HotSpot - Internet - Asterisk - If the
hotspots have ethernet ports, you could go with a PAP2T (2 FXS) and you
could power it with a 12V to 5V converter. Same with a PSTN cordless phone
but I'd be careful
On 11/04/09 18:11, Andrew Hakman wrote:
Did you try changing the FXO port impedance? Did you try resetting the
unit to factory defaults (on the IVR dial 73738# and then 1 to
confirm)? Did you by it from ebay and it shipped from hong kong or
similar? Did you try the linksys silver one on the same
Hi
Is there anyway to add logic to dialplan pattern matching? I would
like to match all toll free numbers with one pattern, so 1800, 1877,
1866, 1855, etc. I can't figure out how to do this in dialplan syntax.
As a programmer, I want to say 18[00 or 77 or 66 or 55 etc]. Can't
figure out if this
What are you reaching out to exactly? It would need to be a Siren14
capable. Also, do you have the Siren codec binary installed? It's not part
of the Asterisk distribution.
Inbound calls to Asterisk work (from a platform that supports both Siren14
and G.711). Leaving ulaw out of the allow
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