Thanks - too bad I missed it :)
2012/11/12 Dan Jenkins dan.jenk...@holidayextras.com
Hi,
As far as I'm aware the videos are still being produced and there's no
definitive list anywhere for the slide decks.
However, my one is here:
On 11/13/2012 12:11 AM, Phil Reynolds wrote:
[snip]
It turns out to be a known issue:
https://issues.asterisk.org/jira/browse/ASTERISK-19532
... and can be fixed by applying the patch at:
https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch
I will file
Hello,
I have the following case.
A customer is a heavy Meetme/audio conference user.
He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference
station).
Users complain they often do not hear the other party loud enough.
The setup is then:
Remote party --- PSTN/ISDN--- Asterisk
Not sure if this is what you want but you can always set the TX and RX
Gain values via the dialplan.
Jacob
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, November 13, 2012 4:46 AM
To: Asterisk Users Mailing
hello,
do you have someone connector to salesforce?
http://wiki.developerforce.com/page/Open_CTI
i need it for Mac OS X (the web/javascript way, not the old CTI Adapter way)
i'm using Asterisk 1.8
thanks
--
---
Marek Cervenka
I have also uploaded my presentation here:
http://www.slideshare.net/alipey/astricon-2012-redundancy-and-high-availability?from=share_email
It's on Redundancy and high availability using OpenSIPS/Kamailio.
Regards,
Ali Pey
On Tue, Nov 13, 2012 at 4:24 AM, Lenz Emilitri lenz.lo...@gmail.com
Hi list,
I'm streaming live mp3 streams (web radios) via mplayer and mpg123 using
MusicOnHold. Often, one or two of the streams die during the day and I
have to restart (not reload) Asterisk to bring it back. moh reload on
the console doesn't do anything.
This is suboptimal if there are
have you tried module reload res_musiconhold.so ?
Cédric
Le 13/11/2012 16:46, Markus a écrit :
Hi list,
I'm streaming live mp3 streams (web radios) via mplayer and mpg123
using MusicOnHold. Often, one or two of the streams die during the day
and I have to restart (not reload) Asterisk to
On 11/12/2012 5:05 AM, Lenz Emilitri wrote:
Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are
available somewhere? I looked at the website but cannot find anything.
Thanks
l.
I'll try to find out today or tomorrow and post the answer on the list.
--
Rusty
In http://packages.digium.com/centos/ there is not yet a centos 6 branch (Nor
is there a RHEL 6 branch). Centos 6.0 was release in July of 2011 - is this
something that Digium is planning on supporting? Or is there a different URL
that I'm not aware of for firmware packages?
-Justin Killen
Am 13.11.2012 16:51, schrieb OCEANET - Cédric BASSAGET:
have you tried module reload res_musiconhold.so ?
Hi Cedric,
thanks for the suggestion. Unfortunately, it does nothing, just like
moh reload.
Any other suggestions?
Regards
Markus
--
module unload res_musiconhold.so
and
module load res_musiconhold.so
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, November 13, 2012 1:00 PM
To: Asterisk Users Mailing List -
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 13, 2012 4:35:54 AM
Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber
session with ejabberd
On 11/13/2012 12:11 AM, Phil
Am 13.11.2012 19:01, schrieb Eric Wieling:
module unload res_musiconhold.so
and
module load res_musiconhold.so
Great, that works, but only if no caller is listening to MOH at that
time. Since *all* my callers are listening to MOH and nothing else, that
means for me it's the same like an
On 11/13/2012 07:05 PM, Michael L. Young wrote:
[snip]
Is it an omission that this fix has not been applied to the 11 tree?
From the looks of ASTERISK-19532 it seems that the fix has only been
applied to 1.8 and 10.
If you click on the link for ASTERISK-19532, there is a tab in the Activity
I have a new install and the customer is complaining that they hear
noise on all calls, no matter if it is internal or external, desk phones
or softphones. The noise is only present when the user is speaking, not
the remote side. The remote side does not hear the noise, only the
local
On 13/11/12 6:52 pm, Carlos Chavez wrote:
Why would a SIP to SIP call have this noise?
Check to see what random stuff they have on their desk.
We've regularly seen things like mobile phones (or cellphones to those
of you across the pond :-) ) causing interference with VoIP phones.
We've
hi,
try to catch in in a cron job per minute.
asterisk -rx 'module unload res_musiconhold.so'
Markus
Am 13.11.2012 19:15, schrieb Markus:
Am 13.11.2012 19:01, schrieb Eric Wieling:
module unload res_musiconhold.so
and
module load res_musiconhold.so
Great, that works, but only if no
Hey, all. It seems that Polycom has a bunch of offerings for
conference phones, and I'm just wondering which are the less-expensive
alternatives; what with their marketing, etc., it's not always obvious
which is which.
Thanks,
-Ken
P.S. If anyone's had really good experience with another
Carlos Chavez cur...@telecomabmex.com writes:
I have a new install and the customer is complaining that they
hear noise on all calls, no matter if it is internal or external, desk
phones or softphones. The noise is only present when the user is
speaking, not the remote side. The remote
Dears;
It seems my service provider is requesting a complicated settings to allow me
to send from behind NAT.
What they said:
It shouldn't matter as long as you are handling the NAT correctly your end. We
do not fix NAT so if you're sending internal addresses in your INVITEs or SDP
then
Dear Bilal,
I understood correctly that the problem is that calls drops?
What router are you using?
Eliezer
On 11/13/2012 11:16 PM, bilal ghayyad wrote:
Dears;
It seems my service provider is requesting a complicated settings to allow me
to send from behind NAT.
What they said:
It
On 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
It seems my service provider is requesting a complicated settings to allow me
to send from behind NAT.
What they said:
It shouldn't matter as long as you are handling the NAT correctly your end.
We do not
It looks like you need to enable the sip application layer gateway or ALG on
your router. The problem is not exactly a Nat issue. The problem is most likely
with the sip header keeping the private IP address, the ALG when enabled will
change this to your public
On 13 Nov 2012, at 21:17, bilal
On 13/11/12 9:31 pm, Leighton Brennan wrote:
It looks like you need to enable the sip application layer gateway or ALG on
your router
Quite often the reverse is true. Most routers (at least those I've used)
seem to have such a lousy implementation of a SIP ALG it's often far
better to just
I'm with Duncan, you need a public IP address, not private. Chris BudinickNetwork Technician
RAINIER CONNECTFrom: "Duncan Turnbull" dun...@e-simple.co.nzTo: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comSent: Tuesday, November 13, 2012 1:29:28 PMSubject:
Check reinvite and NAT settings on the line as well as the SIP peers.
You can use a stun client from inside your network to see what’s going on with
the NAT
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Budinick
Sent:
On 12-11-13 01:16 PM, bilal ghayyad wrote:
Dears;
It seems my service provider is requesting a complicated settings to allow me
to send from behind NAT.
What they said:
It shouldn't matter as long as you are handling the NAT correctly your end. We do
not fix NAT so if you're sending
Carlos,
I think the noise you are hearing might echo cancelation that is broken or set
incorrectly. Maybe the card and asterisk are both trying to echo cancel?
Mark
On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:
I have a new install and the customer is complaining that they hear noise
On 11/13/12 4:31 PM, Mark Engelhardt wrote:
Carlos,
I think the noise you are hearing might echo cancelation that is broken or set
incorrectly. Maybe the card and asterisk are both trying to echo cancel?
Mark
On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:
I have a new install and
Hello All,
Is there a way I can trigger a AGI script On SIP REGISTER event.
--
Any help would be much appreciated.
falazemi
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
Well, an AGI runs in the context of a channel. A REGISTER does not.
So, no.
--
Thanks in advance,
-
Steve Edwards
On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
Well, an AGI runs in the context of a channel. A REGISTER does not.
So, no.
--
Thanks in advance,
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards
Well, an AGI runs in the context of a channel. A REGISTER does not.
So, no.
On Wed, 14 Nov 2012, Face wrote:
Is there a way to accomplish
Hey guys and gals-
Right now, I'm using FreePBX to handle providing voice services to a
handful of customers. However, it just isn't cutting it for features,
billing, customer access (portal stuff), etc. What do you recommend? Is
there an ITSP portal/panel/platform available for running an ITSP
Hello Marshall,
Please see Enswitch:
https://integrics.com/enswitch/
It provides everything you ask for except full IAX support (we recommend
SIP for client connections) and full control of codecs.
It also supports much much more, such as billing, invoicing, payment
collection, etc, etc.
Hey Motty,
The simplest way I've found is having an asterisk console open (asterisk -r)
with verbosity to level 12. Alternatively you could tail -f the full log (in
/var/log/asterisk) - I like to parse it with something like ccze to colour code
things.
The better solution I've found is to use
Hey Dan,
Please keep us updated on a video or transcript of this talk - this seems like
a very fascinating presentation and I'd love to get more information.
Cheers,
Andrew.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan
Dears;
What Jian said is the right and it worked.
But I have the following questions:
Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to
set the localnet or it is enough to set the externip?
From the other side, I am using Asterisk 1.8.12.0 and when I was searching
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