That is correct. That is the first test we did.
On 07/06/2009, Moises Silva moises.si...@gmail.com wrote:
On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote:
Every call as soon as the sangoma card is live.
Speak to Konrad on your techdesk for more info.
Thanks.
I'll
Hello,
I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our
Nortel TX-1 PBX. Up to now I had only the calling party names passed both
ways. After upgrading the Cisco to the latest release (12.4.24T) it began
honoring the remote-part-ID field sent by Asterisk and sends the
Hello,
I have a Sangoma A200 analog card with 2 FXO ports. It's working well
with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk
1.6/dahdi.
I compiled and installed,
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10
wanpipe-3.4.1
asterisk 1.6.1.1
My analog card is recognized
Hello Danni,
As you said, I went through the post and found that is applicable
everytime no matter how many members are there in a conference. And I
understand that I cannot completely rely on that. I need to do some
logical tweaks with some other application like an AGI to crack this
issue. I
Moises Silva schrieb:
On Sat, Jun 6, 2009 at 7:18 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Jose Arias schrieb:
Hi,
Asterisk 1.4.18
AsyncAGI patch from //http://moythreads.com/testasync2.diff
http://moythreads.com/testasync2.diff//
Regards
So what?
What do you mean with so
Hello,
Can anyone give me a sample configuration of Callback feature on a2billing.
Thanks.
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On Sun, Jun 7, 2009 at 5:19 AM, Kurian Thayilkurianmtha...@gmail.com wrote:
extension is hit by channel. I wasn't aware that we can scan a channel
continuously using an AGI. If so, how could we do that?
A somewhat ugly solution that gets the job done:
1) crontab, set up a periodic job that does
On Sun, Jun 7, 2009 at 6:19 AM, Steve Reposcmu...@gmail.com wrote:
I have a Sangoma A200 analog card with 2 FXO ports. It's working well
with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk
1.6/dahdi.
I compiled and installed,
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
On Sun, Jun 7, 2009 at 4:20 AM, Yehavi
Bourvineyehavi.bourv...@gmail.com wrote:
Hello,
I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our
Nortel TX-1 PBX. Up to now I had only the calling party names passed both
ways. After upgrading the Cisco to the latest release
Never mind, it was my mistake. I had some problems with my email client.
Regards
Jose
2009/6/7 Philipp Kempgen philipp.kemp...@amooma.de
Moises Silva schrieb:
On Sat, Jun 6, 2009 at 7:18 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Jose Arias schrieb:
Hi,
Asterisk 1.4.18
When Asterisk sends a call to a phone company via a PRI/Dahdi, does it
actually send CALLERID(ANI), or only CALLERID(NUM)?
Cary Fitch
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Hello all,
I have asterisk-1.4.25, and having some problems with analog lines
becouse my Telco does not have disconnect supervision.
So, i realize there are some switches on main/dsp.c:
BUSYDETECT_MARTIN, BUSYDETECT_TONEONLY, and
BUSYDETECT_COMPARE_TONE_AND_SILENCE.
As far as i could read
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2 files automatically in the end, but this
isn't happening.
I have sox installed in my server.
How can I force Sox
Hi,
I had similar issue which happened when record option was mentioned in
both agents.conf and queues.conf. When I commented the recordagentcalls
option in agents.conf, it started to work. Mention the monitor option
only in the queues.conf file. Do try.
Regards,
Kurian Thayil.
On Sun,
Philipp Kempgen wrote:
sean darcy schrieb:
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
exten =
Thanks Noah for your helpful reply.
My setup will be 2 Asterisk (Trixbox) servers, Active/Passive, 2 PRIs through 2
Vega 400 gateways.
i tried to follow some threats, like as the below:
http://www.trixbox.org/forums/trixbox-forums/open-discussion/ha-cluster
But i think something wrong in
César Sequeira schrieb:
I try to connect Qutecom in my Asterisk Server but without success.
What field I need to complete?
Username;
Password;
Realm (asterisk IP Address);
Default: asterisk
Server (asterisk IP Address);
Proxy (asterisk IP address);
It's correct?
Philipp
Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x
and this is my config:
queues.conf -
[general]
persistentmembers = no
[queue_1]
persistentmembers = no
monitor-format=wav
monitor-join=yes
monitor-type=MixMonitor
wrapuptime=3
Steve Underwood wrote:
Elliot Murdock wrote:
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's
32 bit fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
Use the FAX support built into Asterisk 1.6 and you won't have that
limitation.
Hi Moy,
I'll do it so, but for your answer, it seems you are thinking about it
as it could be a bug. I don't think so. I mean: the redirect action on a
channel in AsyncAGI stops the current agi execution. It's the normal
behavior. It's the way to stop a playfile on a channel if it was
On Sunday 07 June 2009 16:28:53 sean darcy wrote:
Steve Underwood wrote:
Elliot Murdock wrote:
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's
32 bit fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
Use the FAX support built
Tilghman Lesher wrote:
What's the use case for the Digium
driver? Am I missing something by not using it?
While they accomplish the same goal, the commercial driver is based upon
a different codebase,
Ok.
provides support for patented fax protocols,
Really? V.34-fax (33,600 bps) is
Hi all,
I'd like to know the best way to deal with queue member that are reached
trough a SIP trunk. Let me explain:
1) Master asterisk box with my call queue
2) Slave asterisk box with a channel bank interface
the two boxes are connected trough a SIP trunk, and the dialplan in the 1st
box
Jim--
On Thu, Jun 4, 2009 at 1:40 AM, Jim Boykin boykin...@gmail.com wrote:
Hi,
Asterisk does not post CDR when dial status is CHANUNAVAIL.
CDR's are, at the current time, and always have been attached to the channel
struct;
so, if you don't create a channel, then there is nowhere to
On Sun, Jun 7, 2009 at 4:37 PM, Jose Ariascyr2...@gmail.com wrote:
Hi Moy,
I'll do it so, but for your answer, it seems you are thinking about it as it
could be a bug. I don't think so. I mean: the redirect action on a channel
in AsyncAGI stops the current agi execution. It's the normal
On Sunday 07 June 2009 19:39:50 Lee Howard wrote:
Tilghman Lesher wrote:
What's the use case for the Digium
driver? Am I missing something by not using it?
While they accomplish the same goal, the commercial driver is based upon
a different codebase,
Ok.
provides support for
Tilghman Lesher wrote:
On Sunday 07 June 2009 19:39:50 Lee Howard wrote:
Tilghman Lesher wrote:
What's the use case for the Digium
driver? Am I missing something by not using it?
While they accomplish the same goal, the commercial driver is based upon
a different
I have 2 hosts that Asterisk is in between of...and for both I have
canreinvite=no - but asterisk still sends re-invite to get out of the media
path.
Proxy 1 -- Asterisk-- Proxy 2
I want asterisk to anshor media..
In extenstions.conf I have an entry to send calls for say 5551000 to proxy 2
On Fri, May 29, 2009 at 11:35 PM, Olivier oza-4...@myamail.com wrote:
2009/5/29 Danny Nicholas da...@debsinc.com
I’m pretty sure that attended transfer is a “features” function, not a
dialplan one.
Yes, you're right but do you think there's such a big difference between
both that it
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