Yes I'm aware of g711ulaw and PCMU..but I don't want to use this codec in
Phones, as I'm using more then 10 IP phones in my network and going to increase
it to connecting other remote offices,, and I don't want to increase the
Bandwidth usage due to g711ulaw..
Keshav
Jared Smith
Yes I've testes by this also as only allowing g719 codec, but in that case
asterisk is sending 488 Not acceptable here, because INVITE form the phone is
having g711ulaw and g711alaw
Keshav
dave cantera [EMAIL PROTECTED] wrote: keshaw,
did you set your sip.conf to only allow g729?
hi,
thnks 4 reply,
actully i am using asterisk 1.4.15 and that is defined in menuselect
file.(xml file)
so no need to add entry in module.conf
Bhrugu mehta
On Dec 27, 2007 7:37 PM, dave cantera [EMAIL PROTECTED] wrote:
bhrugu,
did you try and load it manually?
Modules are compiled in to
On Dec 28, 2007 12:08 AM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
I'm assuming that since you sent it to Asterisk Users (Non-Commercial
Discussion) it is free.
Is it also Open Source?
Classic Matt! And also (to Matt) Classic, Matt!
Grey Man wrote:
- Original Message
From: Steve Murphy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, 27 December, 2007 5:44:01 PM
Subject: Re: [asterisk-users] CDR
Greyman--
No real new
Hey everybody,
I've just spent the last two hours Googling and searching the Wiki. I'm
trying to find if there are any listings of codes for the Avaya Definity
G3R, to allow for an Asterisk system to turn on/off a phones MWI that is
attached to a G3. We are looking to use an Asterisk system
I've just spent the last two hours Googling and searching the Wiki. I'm
trying to find if there are any listings of codes for the Avaya Definity
G3R, to allow for an Asterisk system to turn on/off a phones MWI that is
attached to a G3. We are looking to use an Asterisk system as a voice
Hi all,
sorry to rehash this - but I'm having similar issues. I'm on Asterisk 1.0
and have been using Queues without any problems locally. I mean, all the
SIP devices on my local server can be added to queues using AddQueueMember.
However, I now need to allow agents from other servers to log in
Grey Man wrote:
On a separate note does anyone know how to block transfers on a SIP
channel? I can block REFER requests from my SIP Proxy but I have to
support some transfers so that's not an option.
I'd put the SIP devices in a separate context that doesn't include any
[twkTWK] in the Dial
Dean,
I am saying nothing of the sort.
To clarify, I am saying that I do not see the people you mentioned
fishing for free ideas or posting commercially to the User's list with
the exception of yourself now, when you had affiliation with Mexuar, and
a handful of other times.
I find it
Hi list,
Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run
into some serious problems. The first thing I noticed was this message
that would show up every five seconds on the CLI:
Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No
D-channels
I have been playing with this some time ago. We used the so called mode
code integration. This worked fine. It works simular as described for
other Avaya Product.
http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration
Henk
BJ Weschke schreef:
I've just
On 12/27/07, broadband Voice [EMAIL PROTECTED] wrote:
I am using Asterisk and A2billing Calling Card Platform and after the 6th
call the quality starts to degrade. The way it set up is the user calls into
the system then dial out so I have 12 channels being used up but 6 active
calls. Here
Dear all
I've got call queuing working when calls originate from my local site.
After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a
BJ Weschke wrote:
I think you're going to need to integrate via the SMDI feature of
Asterisk and figure out what the Definity needs as well to work with an
SMDI connection.
Thanks for the input.
Doug
--
Ben Franklin quote:
Those who would give up Essential
Henk Dick wrote:
I have been playing with this some time ago. We used the so called mode
code integration. This worked fine. It works simular as described for
other Avaya Product.
http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration
Yes, I saw the page.
broadband Voice wrote:
On 12/27/07, *broadband Voice* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am using Asterisk and A2billing Calling Card Platform and after
the 6th call the quality starts to degrade. The way it set up is
the user calls into the system then dial
Doug,
Have you checked the feature access code that is defined in the
definity. That is the code that needs to be dialed. I always checked
the codes from a definity phone to make sure that I was using the right
codes.
Henk
Doug Lytle schreef:
Henk Dick wrot
I have been playing with
Henk Dick wrote:
Doug,
Have you checked the feature access code that is defined in the
definity. That is the code that needs to be dialed. I always checked
the codes from a definity phone to make sure that I was using the right
I have not been able to find any references to the
You're looking for Leave Word Calling activation and deactivation.
On 12/28/07, Doug Lytle [EMAIL PROTECTED] wrote:
Henk Dick wrote:
Doug,
Have you checked the feature access code that is defined in the
definity. That is the code that needs to be dialed. I always checked
the codes
Hi list.
I'm new to IVRs and trying to set up one that toggles an auto-forward
flag on or off for specific accounts.
I'd like to have my users dial an extension and then be prompted to
enter the account number. (done)
Next I'd like it to jump to the appropriate line in the dial plan that
Hi -
I'm looking into realtime and I'm having a bit of a problem with the SIP
part.
My review of the posts seems to indicate that I should use realtime static
for the [general] part of my sip.conf including the registration commands:
register=did:secret@domain/did context
and use realtime
It's licensed GPL. I'm working on getting the web-site, documentation, and
packaging up to par... if you're interested in helping, let me know.
Here are some details on it:
* Written for Asterisk 1.4.x; not tested with prior versions
* Supports both voice and fax mail (including fax detection)
On Thu, 27 Dec 2007, broadband Voice wrote:
I am using Asterisk and A2billing Calling Card Platform and after the 6th
call the quality starts to degrade. The way it set up is the user calls into
the system then dial out so I have 12 channels being used up but 6 active
calls. Here are my specs
On Fri, 28 Dec 2007, Steve Totaro wrote:
broadband Voice wrote:
On 12/27/07, *broadband Voice* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am using Asterisk and A2billing Calling Card Platform and after
the 6th call the quality starts to degrade. The way it set up is
This system targets a different market...
I like Olle's system. He did a good job. Olle's minivm is a great choice for
those wishing to build customized voicemail systems, but as the name suggests,
the systems are very basic.
Large systems are difficult to maintain in the dial plan and some
I know a lot of people on this list are building devices and equipment for
Asterisk and communications in general...
For those of you building prototype devices, you may want to check out TechShop
in the bay area. They are expanding all over the place.
http://www.techshop.ws
They have
Gordon Henderson wrote:
On Fri, 28 Dec 2007, Steve Totaro wrote:
broadband Voice wrote:
On 12/27/07, *broadband Voice* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am using Asterisk and A2billing Calling Card Platform and after
the 6th call the quality starts to
Red Tiger is Java based, so it will run on any Java VM (i.e., Windows, MacOS,
Linux, Unix, etc.)
There are some JNI-based additions for Linux which give it more capabilities,
but Red Tiger itself runs cross platform.
You could run Asterisk on Linux, but have Red Tiger and all of your
Tom Lynn wrote:
You're looking for Leave Word Calling activation and deactivation.
Thank you, I'll pass that on to him.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Philipp Kempgen wrote:
Drew Gibson wrote:
A well-written application should attempt to minimize the amount of
'conversion' the user/programmer has to do. Therefore the command
structure SHOULD be in a form that is natural for the user/programmer,
NOT to the machine.
Personally, I would
I figured it out. The ftp site was not named well and corrected. The other
problem I have it after the extraction and make; it was suppose to go under
/etc but that did not happen. I am trying to figure out why.
On 12/28/07, broadband Voice [EMAIL PROTECTED] wrote:
I successfully downloaded the
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
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Al lists wrote:
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
I think that you mean to say HylaFAX and IAXmodem or txfax/rxfax ...
because spandsp is but a DSP/DCE library, and it cannot work alone, and
iaxmodem uses spandsp.
Thanks,
Lee.
Al lists wrote:
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
HylaFAX+ and iaxmodem (That includes SpanDSP).
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Hi list,
Just thought I'd let you know that the problems outlined in my
previous post apparently had to do with a bad card. After swapping it
out for another one
the messages went away.
Of course, I still have some problems. For instance, there's this
error that keeps appearing in my
So HylaFax and IaxModem is more preferred than using rxfax/txfax ?
any reason?
On Dec 28, 2007 6:40 PM, Lee Howard [EMAIL PROTECTED] wrote:
Al lists wrote:
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
I think that you mean to say HylaFAX and IAXmodem or
On Sat, Dec 29, 2007 at 02:46:18AM +0100, Jaap Winius wrote:
Hi list,
Just thought I'd let you know that the problems outlined in my
previous post apparently had to do with a bad card. After swapping it
out for another one
the messages went away.
Of course, I still have some
On Fri, Dec 28, 2007 at 07:56:39PM -0500, broadband Voice wrote:
I figured it out. The ftp site was not named well and corrected. The other
problem I have it after the extraction and make; it was suppose to go under
/etc but that did not happen. I am trying to figure out why.
On 12/28/07,
Hi Justin,
On Thu, 2007-12-27 at 15:38 -0800, Justin Newman wrote:
Yes, I wrote nvfaxdetect and a number of other modules. I don't have
any nvfaxdetect updates planned for public release unless someone
would like to integrate some of my changes in the GPL version...we
could do this though.
On Fri, 28 Dec 2007, Steve Totaro wrote:
Gordon Henderson wrote:
On Fri, 28 Dec 2007, Steve Totaro wrote:
broadband Voice wrote:
On 12/27/07, *broadband Voice* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am using Asterisk and A2billing Calling Card Platform and after
Does anyone know how much space the appliance has for voicemail and/or logs?
Doesn't have an embedded disk from what I can see, and only a 1G flash card?
--
Barry D. Hassler
President, HCST
http://www.hcst.net/
937-427-9000
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