From what I understand your sip client can handle g729 whereas for DAHDI you
need transcoding to a|ulaw.
I am using it with no problems (have g729 licenses as well though).
A bit off topic, I have found some extra configuration that is not really in
the docs (or I could not find them):
I am using asterisk 1.6 at home and would like to send incoming calls
without caller id immediately to voicemail (i don't want to use the
privacy manager where people have to enter a number).
The config examples i found are all for the pretty obsolete 1.0 and 1.2
versions of asterisk.
Would
That's my point - SFA comes with a g729 licence, so why can't it
transcode to the DAHDI channel ?
Thanks also for the info. Very useful.
Julian
2009/12/6 Roeften roef...@gmail.com:
From what I understand your sip client can handle g729 whereas for DAHDI you
need transcoding to a|ulaw.
I am
On 6 Dec 2009, at 08:56, Remco Barendse wrote:
I am using asterisk 1.6 at home and would like to send incoming calls
without caller id immediately to voicemail (i don't want to use the
privacy manager where people have to enter a number).
The config examples i found are all for the pretty
James you are right. let me add one more line
exten = s,n,GotoIf($[${CALLERID(num)}=]?nocid,s,1)
On Sun, Dec 6, 2009 at 3:18 PM, James Stocks stoc...@stocksy.co.uk wrote:
On 6 Dec 2009, at 08:56, Remco Barendse wrote:
I am using asterisk 1.6 at home and would like to send incoming calls
Interesting response but I am not that saavy to follow it!
Thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
And, then send an email to the party. Example
3035551...@tmobile.net
Thomas Perron wrote:
Interesting response but I am not that saavy to follow it!
Thank you
Then I'd suggest you hire someone that is more technically inclined to
help you with your project.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Thomas Perron wrote:
Interesting response but I am not that saavy to follow it!
Thank you
If you can't follow his response, perhaps you need to hire someone to
write your dialplan?
I know next to nothing regarding coding, and nothing of 1.6, and I
understood it.
He practically wrote it
Hello,
I'm trying to figure out how to limit the number of concurrent calls a client
can make.
I have a client that has 6 SIP accounts. One for each SIP phone.
I want to limit it so that they can only make 2 outgoing calls at a time so
that I can bill them per channel rather than per
Julian Lyndon-Smith wrote:
That's my point - SFA comes with a g729 licence, so why can't it
transcode to the DAHDI channel ?
It comes with a license, but does not include the transcoding
functionality itself. You need to download and install the appropriate
Digium codec_g729 module for your
I am trying to use a simple tool in the Dial plan so that if the first
number does not connect the logic will go to the second and/or third.
Basically, I want the call to ring and connect to the first number
Then, if it is not answered I want another number to try to get connected
Then, if second
Zeeshan Zakaria schrieb:
I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk
1.4 using realtime architecture. Extensions
extensions == sip.conf peers?
are defined in realtime database
and dial plan is in AEL. I am able to correctly setup hints in the dialplan,
but
Gayathri G schrieb:
1. Can I use Asterisk as a SIP Proxy. ( I want it to act as proxy not a
B2b/GW)
No. Asterisk is a back-to-back user agent (B2BUA).
You might want to have a look at
http://en.wikipedia.org/wiki/OpenSER
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied
Hallo,
ABCTI (an open-source CTI client for Asterisk) has moved to beta stage.
Find it on:
http://abcti.sourceforge.net
For the first time, we now have windows installers that actually work ;-)
We would appreciate any feedback you can give.
Regards,
-- o
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application generally
echo body of message .?
mail -s --what does this
Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on the
other one with Asterisk 1.4.18. Both have exact same sip.conf and
extensions.conf, same extension numbers. Is there anything else which could
effect it. The one on which it doesn't work is on a virtual machine, on a
Try Googling some of this stuff, such as
linux mail -s
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Thomas Perron schrieb:
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application generally
echo body of message .?
Aha. That was it. Thanks.
I could not see that advice in the documentation. I may be blind, but
it may be helpful to include it somewhere.
Thanks again
Julian
2009/12/6 Kevin P. Fleming kpflem...@digium.com:
Julian Lyndon-Smith wrote:
That's my point - SFA comes with a g729 licence, so why
Zeeshan Zakaria schrieb:
Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on the
other one with Asterisk 1.4.18. Both have exact same sip.conf and
extensions.conf, same extension numbers. Is there anything else which could
effect it. The one on which it doesn't work is on a
On Sun, Dec 6, 2009 at 12:28 PM, Thomas Perron thomas.per...@gmail.com wrote:
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application
Hi all,
we have
this annoying problem with Linksys SPA9x2 phones and echo cancellation. I have
read
posts on other sites about this problem but they are more than one year old and
people were using older firmware. Linksys/Cisco has released 6.1.5a firmware
but we still experience the same
Julian Lyndon-Smith wrote:
Aha. That was it. Thanks.
I could not see that advice in the documentation. I may be blind, but
it may be helpful to include it somewhere.
Please open a ticket with Digium's support department documenting the
issues you ran into setting up the product; that way it
Andrew Witt wrote:
I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with
recording CDRs using MySQL. Unlike all of the other postings and web
pages I have found on this issue, my installation successfully stores
the -first- CDR, but nothing after that.
It looks like
Actually it was the 'call-limit' option which needed to have some value more
than 0. By default the realtime extensions have it set to 0, and there was
no such field in `sip-buddies` table. So I created it in `sip_buddies` and
set it to 50, and now hints work just perfectly fine as they should.
is the realm value used in authentication proccess is case sensitve?___
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On 7/12/09 8:11 AM, Dubravko Caric wrote:
Hi all,
we have
this annoying problem with Linksys SPA9x2 phones and echo cancellation. I
have read
posts on other sites about this problem but they are more than one year old
and
people were using older firmware. Linksys/Cisco has released 6.1.5a
Hi Friends,
need to help.
*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*
Scenario is like as :
-
1) kamailio server on
Would it not be easier for you to just bill them for access to 12 channels (6
extensions x 2 channels each)? Seems simpler. Then bill them for the calls
they actually make.
Then set call-limit=2 for each extension in sip.conf?
See:
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