Hello all,
I have a problem of Music on Hold on AsteriskNow system, based on Asterisk
1.6.2.19 with FreePBX 2.8.1.4
On another system, when we press the HOLD button on the phone, the phone
sends an INVITE with a=sendonly in the SDP, and we get an OK and the system
recognizes the a=sendonly
Hello,
We installed AsteriskNOW (Asterisk version 1.6.2.19) and purchased FFA and
G.729 licenses and installed them successfully.
When we tried to install asterisk addons (we need the CDRs), we get license
conflict error messages.
Searching google, we see that there's a way to force install the
Am 18.07.11 16:15, schrieb Alex Vishnev:
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA
on another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer
Been looking at SwitchVox and how it handles mobility using virtual extensions.
Does somebody have any examples on how this can be achieved with Asterisk ? I
have Bria on my Android and it would be nice if I could get my office phone
and/or cell to ring.
--
Thanks, Phil
--
On 19/07/11 08:20, Michael wrote:
On the AsteriskNow system, it gives an OK, but nothing happens, there's
no music and after some time, the call even drops for empty RTP. That's
the log there:
What does the Asterisk CLI show when this happens on your AsteriskNow
system?
--
Hi All;
I succeeded now to configure callpickup so if the SIP user pressed *8 then it
will pickup the call within the group.
What is the possibility to have another code (for example *7 or any thing else)
to pickup the call from another callgroup, for example: if I pressed *8 then I
can
No, that looks like a separate issue. Mine is a 100% repeatable and the
asterisk does not lock up. SIP and RTP on other sessions are still going. in my
cases this is the exchange I see
Asterisk
On 07/18/2011 05:05 PM, Elliot Murdock wrote:
I am wondering if the Libss7 add-on for Asterisk also translates ss7
variables into the dialplans for routing, accounting, etc?
What are 'ss7 variables'?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber:
On 07/19/2011 02:24 AM, Michael wrote:
Hello,
We installed AsteriskNOW (Asterisk version 1.6.2.19) and purchased FFA
and G.729 licenses and installed them successfully.
When we tried to install asterisk addons (we need the CDRs), we get
license conflict error messages.
You don't need to
Michael,
Here are the differences between the systems that I determined from the two SIP
traces:
* Working system: no NAT, phone codec: G.729, Asterisk codec: G.729
* Non-working system: NAT, phone codec: G.729, Asterisk codec: A-law
Does the conversation have two-way audio prior to the
Hi,
Is there a easy way to configure the sip settings so it is not possible to
register more than one sip user with the same username/password.
Now it is possible to register more than one sip user with the same
username/password.
So if I call that sip user, both sip clients will ring.
On 07/19/2011 07:50 AM, Arjan Kroon | Mobillion wrote:
Is there a easy way to configure the sip settings so it is not possible to
register more than one sip user with the same username/password.
Now it is possible to register more than one sip user with the same
username/password.
So if I
On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
/usr/lib/asterisk/modules/
Be sure to only include the ones you need. Finding which exactly may be
tricky.
Thanks Tzafrir. Actually, since the modules are the biggest files by
far, besides the obvious (SIP, Dahdi,
On Mon, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote:
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this,
On Mon, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote:
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen
Sent: Tuesday, July 19, 2011 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [1.4] Minimal installation?
On Mon,
Hi,
On Tue, 2011-07-19 at 16:14 +0200, Gilles wrote:
On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
/usr/lib/asterisk/modules/
Be sure to only include the ones you need. Finding which exactly may be
tricky.
Thanks Tzafrir. Actually, since the modules
On Tue, 2011-07-19 at 03:23 -0700, bilal ghayyad wrote:
Hi All;
I succeeded now to configure callpickup so if the SIP user pressed *8 then it
will pickup the call within the group.
What is the possibility to have another code (for example *7 or any thing
else) to pickup the call from
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes to
be working togethor as one entity so a distributing for the load and for the
tasks can be acheived.
I need such kind of protocols to be used in a large call center is where alot
of E1s and alot of agents.
On Tue, 2011-07-19 at 08:53 -0700, bilal ghayyad wrote:
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes
to be working togethor as one entity so a distributing for the load and for
the tasks can be acheived.
I need such kind of protocols to be used in
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes to
be working togethor as one entity so a distributing for the load and for the
tasks can be acheived.
I need such kind of protocols to be used in a large call center is where alot
of E1s and alot of agents.
Dear;
Thanks for your help.
And in that case, if will be considered logged in at the extension that he
dialed from it the code to login as agent, correct? So, the agent will receive
the calls at that extension?
By the way: When I did this, it did not ask me for the agent password !! So,
when
On Tue, 2011-07-19 at 09:24 -0700, bilal ghayyad wrote:
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes
to be working togethor as one entity so a distributing for the load and for
the tasks can be acheived.
I need such kind of protocols to be used in
On Tue, 19 Jul 2011 09:27:41 -0500, Danny Nicholas
da...@debsinc.com wrote:
My .02 - FWIW, DAHDI will use almost as much space as the rest of Asterisk,
so you could save the space you don't have by forgoing that.
Thanks everyone for the feedback. I'll go through the list of modules
and see what I
Hello All,
I have asterisk server running on Centos, some of our users are spreadout
throut the states. I want the time zone to reflect our users repective time
zones. My questions is how to customize their timez zone accordingly? Is
that done in sip.conf? or extensions.conf?
Thanks in
motty.cruz would like to recall the message, Time zone on phones.
attachment: winmail.dat--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
motty.cruz wrote:
motty.cruz would like to recall the message, Time zone on phones.
Good luck on that one! Exchange feature only.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
On Tue, Jul 19, 2011 at 12:36 PM, Paul Hayes p...@provu.co.uk wrote:
What does the Asterisk CLI show when this happens on your AsteriskNow
system?
Absolutely nothing, unlike the working system, where the CLI clearly
indicates that MOH is activated
--
On 07/19/2011 01:07 PM, Michael wrote:
We would like Asterisk to listen on port 5060 and on an additional port.
From what we read online, it's not really possible, so is it possible to
install a separate instance of Asterisk on the same machine (without
using Vmware or such) and set the 2nd
On 07/19/2011 02:15 PM, Kevin P. Fleming wrote:
Actually, you can do this with one installation of Asterisk, and a
separate set of config files and data directories. When the Asterisk
executable is started, the '-C' option can be used to point to an
asterisk.conf file; that file can then tell
On 07/19/2011 01:16 PM, Alex Balashov wrote:
On 07/19/2011 02:15 PM, Kevin P. Fleming wrote:
Actually, you can do this with one installation of Asterisk, and a
separate set of config files and data directories. When the Asterisk
executable is started, the '-C' option can be used to point to an
I prefer
How do we do that? Isn't Asterisk a SIP Proxy ;)?
That's a good question...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, July 19, 2011 2:18 PM
To:
On 7/19/2011 2:07 PM, Michael wrote:
We would like Asterisk to listen on port 5060 and on an additional port.
From what we read online, it's not really possible, so is it possible to
if you're running iptables, you can set up a pretty simple rule to
forward your additional port to 5060.
On 07/19/2011 02:25 PM, Jeremy Kister wrote:
On 7/19/2011 2:07 PM, Michael wrote:
We would like Asterisk to listen on port 5060 and on an additional
port.
From what we read online, it's not really possible, so is it
possible to
if you're running iptables, you can set up a pretty simple rule
On Tue, Jul 19, 2011 at 9:14 PM, Alex Balashov abalas...@evaristesys.comwrote:
You'd use something from the OpenSER/Kamailio/OpenSIPS family. Those are
actual proxies.
We'll try that, if it's not too complicated. Thanks.
--
On 07/19/2011 01:02 PM, Michael wrote:
On Tue, Jul 19, 2011 at 3:34 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
You don't need to install asterisk-addons to be able to store CDRs; you need
them to be able to store CDRs in MySQL specifically. If you
Michael wrote:
True. In the working system, LAN calls are also using G.729, while
in the non-working system, LAN calls are in G.711 (supported but
not prioritized by the phones) and only the SIP trunk to the ITSP
is set to G.729.
Can you set the phone to G.711 and try making a LAN call on
On Tue, Jul 19, 2011 at 11:53 AM, motty.cruz motty.c...@gmail.com wrote:
Hello All,
I have asterisk server running on Centos, some of our users are spreadout
throut the states. I want the time zone to reflect our users repective time
zones. My questions is how to customize their timez zone
On Tue, Jul 19, 2011 at 9:49 PM, Jason Parker jpar...@digium.com wrote:
Yes, but you don't have to use cdr_mysql to insert into a MySQL database.
The cdr_odbc module works just fine for that.
So what's the procedure required to set FreePBX CDRs active, under these
conditions? How do I
Hello,
Is SS7 and PRI in any way compatible in that if the interface is
configured one it will work for the other (granted, it will not have
any of the ISUP, etc. parameters available if the line is PRI) or are
they two distince protocols that have incompatible signalling?
Thanks,
Elliot
--
On 07/19/2011 02:48 PM, Elliot Murdock wrote:
Is SS7 and PRI in any way compatible in that if the interface is
configured one it will work for the other (granted, it will not have
any of the ISUP, etc. parameters available if the line is PRI) or are
they two distince protocols that have
Hello Kevin,
SS7 parameters to Asterisk variables.
--Elliot
On Tue, Jul 19, 2011 at 3:31 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 07/18/2011 05:05 PM, Elliot Murdock wrote:
I am wondering if the Libss7 add-on for Asterisk also translates ss7
variables into the dialplans for
On 07/19/2011 03:11 PM, Elliot Murdock wrote:
SS7 parameters to Asterisk variables.
In that case, yes, various SS7 values/parameters will be available as
Asterisk channel variables. They aren't documented much, unfortunately,
but you can see them in channels/sig_ss7.c; just search for
Hi,
On Tue, 2011-07-19 at 14:30 -0500, Warren Selby wrote:
On Tue, Jul 19, 2011 at 11:53 AM, motty.cruz motty.c...@gmail.com
wrote:
Hello All,
I have asterisk server running on Centos, some of our users
are spreadout
throut the states. I want the time zone to
On 2011-07-19 20:07, Michael wrote:
Hello,
We would like Asterisk to listen on port 5060 and on an additional
port. From what we read online, it's not really possible, so is it
possible to install a separate instance of Asterisk on the same
machine (without using Vmware or such) and set the
Hi,
so is it possible to install a separate instance of Asterisk on the same
machine (without using Vmware or such) and set the 2nd instance to
listen on another port?
you might try it with Xen as virtualisation solution, because especially
when using it with Linux as paravirtualised guest, it
On 07/19/2011 04:33 PM, Thorolf Godawa wrote:
Hi,
so is it possible to install a separate instance of Asterisk on the same
machine (without using Vmware or such) and set the 2nd instance to
listen on another port?
you might try it with Xen as virtualisation solution, because especially
when
On 7/19/11 11:07 AM, Michael wrote:
We would like Asterisk to listen on port 5060 and on an additional port. From
what
we read online, it's not really possible, so is it possible to install a
separate
instance of Asterisk on the same machine (without using Vmware or such) and set
the 2nd
I can confirm as well that there is an issue with Asterisk crashing.
Asterisk 1.6.2.19 was installed using Digium repository. Probably some
module was enabled in the repository install that is causing this.
On Mon, Jul 18, 2011 at 12:13 PM, Lee Archer lee.arc...@thebigword.comwrote:
Hi Kevin,
Hi Arian,
Now it is possible to register more than one sip user with the same
username/password.
So if I call that sip user, both sip clients will ring.
I've been able to accomplish this with mixed results. In most cases on my
end, the SIP that was most recently registered is the one that
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