http://vuc.me
Kamailio, Open SER and Asterisk walk into a bar...
The bartender is Alex Balashov, someone whose posts I have long
admired on this list. Alex has agreed to take us through the following
areas:
- Relationship of Kamailio to OpenSER project history.
- What is Kamailio/OpenSER?
- S
If anybody who want to earn a quick $50 via paypal and can help me on
setting up a polycom ip7000 to work with asterisk please email sam __ tam AT
hotmail DOT com
Do not email me through my gmail account.
Sam
___
-- Bandwidth and Colocation Provided
On Dec 17, 2009, at 10:36 PM, Neeraj Chand wrote:
>
> Just finished with the instructions from digium website/ net on how to
> compile FFA:
>
> After restart, modules did not get loaded so tried to load manually:
>
> [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module:
> Error l
Just finished with the instructions from digium website/ net on how to
compile FFA:
After restart, modules did not get loaded so tried to load manually:
[Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module:
Error loadin ile: No such file or directory
[Dec 18 14:31:26] WARNING[1100
On Fri, Dec 18, 2009 at 12:23:22AM +0100, Olivier wrote:
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I can't
> nicely test the 9am or 2pm branch.
>
> Suggestions ?
How about setting, say, LUNCHTIME to "
I'm trying to figure out how calls are ending up in my default
context (which should never happen).
I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip
phones.
When I make a call from one of the FXS ports on the 1760, the call
goes into asterisk's default context instead of wher
On Thu, Dec 17, 2009 at 6:23 PM, Olivier wrote:
> Hi,
>
> When I was testing an IVR, I realized I miss a function I would call
> GotoIfTimeWithOffset.
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I can't
>
On Fri, 18 Dec 2009, Olivier wrote:
> Today, this IVR is using function AEL GotoIfTime in several places. The
> problem is if it's 11pm at the moment I'm testing this IVR, I can't
> nicely test the 9am or 2pm branch.
Wouldn't a "set time" function be more usefull?
You could set the time in a s
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Ma
At 12:23 AM on 18 Dec 2009, Olivier wrote:
> Hi,
>
> When I was testing an IVR, I realized I miss a function I would call
> GotoIfTimeWithOffset.
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I
> can't n
On Thursday 17 December 2009 17:23:22 Olivier wrote:
> When I was testing an IVR, I realized I miss a function I would call
> GotoIfTimeWithOffset.
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I can't
> nic
Hi,
When I was testing an IVR, I realized I miss a function I would call
GotoIfTimeWithOffset.
Today, this IVR is using function AEL GotoIfTime in several places.
The problem is if it's 11pm at the moment I'm testing this IVR, I can't
nicely test the 9am or 2pm branch.
GotoIfTimeWithOffset would
Is it certain the issue relates to vserver ?
Maybe it wouldn't work either without vserver ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.dig
On 12/15/09 12:30, VoIP Newbie wrote:
>Joseph,
>You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
>provides real FXO port that registers with Asterisk.
>David
Are you using it?
I could not find much information bout it on wiki; besides I think I'll settle
for Audiocodec 114 (no
Scott L. Lykens wrote:
> I don't mean to be rude in calling you out about it, however, I've been
> waiting for three months for an appearance of Fax for Asterisk that is
> compatible with 1.6.1.5+. Several previous requests for timelines to
> this list have resulted in responses indicating it was
Yes, It happens also, I am having this problem in many servers with
different boards, asterisk versions and drivers (zap and dahdi). Sometimes
in one month it is all fine in the other I get a lot of this off hook
states.
2009/12/17 Danny Nicholas
> It could be a “bad” wire from the fax to th
It could be a "bad" wire from the fax to the TDM card that registers as
"off-hook". If you move the fax to a different port on the card, does the
condition follow it?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandr
Hello Tzafrir Cohen,
First of all thanks for the reply. :)
The fax normally works. But two or tree times a month it goes to this state.
One time, with the state offhook, I disconnect the fax line and connected a
phone. The result was the same, it stays offhook.
Can it be some strange electrical
Hello all,
I'm looking for some help to try to understand why my CPE doesn't work
good with Asterisk in MGCP.
Here is what I want to do :
- Register a TECOM AH4021 on Asterisk in MGCP with the following profile
in mgcp.Conf :
[general]
port = 2727
bindaddr = 10.95.20.1
disallow=all
allow=g729
a
On Fri, Dec 04, 2009 at 03:33:00PM +, Alexandre Rodrigues wrote:
> Hello again,
>
> Adding more information:
>
> Core show channels:
>
> Channel Location State Application(Data)
> DAHDI/4-1s...@national_mobile:1 Rsrvd(None)
> DA
Has anyone an idea of why is this happening
The problem usually happens when I have a fax connected to a fxs line. I
tried many "adaptation", of the dahdi configuration, but the problem
persists.
I am starting to think I have a hardware problem with the Digium Board.
Can anyone recommend me
> It is frustrating to me as we are encouraged to upgrade due to security
> issues but if we want to use this particular Digium product we cannot. I
> have chosen to upgrade as we have not purchased Fax for Asterisk and as
> we are unable to evaluate it I doubt we will. (Not to be snarky but I
> do
yes, sox is installed.
Anyway, I changed the lines that read: Monitor(gsm,/var/log) to
MixMonitor(/var/log/file.gsm...)
Thanks for answering.
2009/12/16 Holger von Ameln
> This may be pretty obvious but do you have sox installed? I managed to
> forget that on more than one occasion ;-)
>
>
> Subject: Re: [asterisk-users] Mixing commercial/SVN Asterisk
>
> On Dec 16, 2009, at 10:08 AM, Richard Kenner wrote:
>
> > Am I correct that if I'm running an -rc or from an SVN release tree
> > that there's no way I can use any commercial add-ons from Digium,
such
> > as Skype, Cepstral, or G.
On Tue, Dec 15, 2009 at 07:53:47PM +, Jeff LaCoursiere wrote:
>
> On Tue, 15 Dec 2009, Ben Schorr wrote:
>
> > O.K., interestingly enough when I call our extensions from my mobile
> > phone it still seems to be using ULAW, but when they dial out it seems
> > to be using G.729 now.
> >
> > Is
> I think you need to remove the line echocanceller in system.conf
> You could also try to use fxotune, it'a really improving things.
> You also need to put echocancel=yes in chan_dahdi.conf
This is a PRI, so fxotune is not the thing to use in this case.
- Noah
_
On Thursday 17 December 2009 04:37:52 Landy Landy wrote:
> --- On Wed, 12/16/09, Landy Landy wrote:
> > From: Landy Landy
> > Subject: Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > Date: Wednesday, December
--- On Wed, 12/16/09, Landy Landy wrote:
> From: Landy Landy
> Subject: Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Date: Wednesday, December 16, 2009, 7:28 AM
> > peering useful: http://astrecipes.net/i
Hi,
I have just installed asterisk, I want to send registration request to
192.168.4.3:6090 and the domain should be test1.net
I have added the following line to sip.conf
register => 897...@test1.net:pazzwrd:897...@192.168.4.3:6090
now the problem is that the SIP Request is appearing as 192.168.
Dear,
some iax phones,(with built in router) have problem, with our asterisk
server, there is no way sound if they call out, but it's ok if somebody
calls them.
the normal iax phones without router have'nt ny problem.
can u help me?
the version of kernel is 2.6.18 and asterisk is 1.4.26.2
Best
Dear,
some iax phones,(with built in router) have problem, with our asterisk
server, there is no way sound if they call out, but it's ok if somebody
calls them.
the normal iax phones without router have'nt ny problem.
can u help me?
the version of kernel is 2.6.18 and asterisk is 1.4.26.2
Best
Friends,
At the first Astricon I was very happy to see Marc Blanchet as one of the
attendees. I knew he was one of the IPv6 gurus and wanted someone to show some
interest in Asterisk and IPv6.
Well, he did not only get interested in it, but started coding on it. The
results have been availabl
32 matches
Mail list logo
