Hi everybody,
I've been working on a project which records the voice of the incoming call.
I use record_file function of asterisk as described below:
RECORD FILE filename format escape digits timeout [offset samples]
[BEEP] [s=silence]
filename: record1
format: wav
escape digits: #
timeout: -1
Hey,
How are you starting the recording? MixMonitor? or Monitor? or some option
in an application?
If you are using MixMonitor or anything alike then you should
StopMixMonitor when the call hits the h extension.
Paste your dialplan relevant to the recording scenario to suggest you
something
Hi Sammy,
Thank you for the reply.
My dialplan is as follows:
exten = 500,1,Answer()
exten = 500,2,Playback(wellcome); play the wellcome message
exten = 500,3,AGI(recording.pl) ; Do the echo test
exten = 500,4,Playback(demo-echodone) ; Let them know it's over
exten = 500,5,Hangup
2011/11/3 Danny Nicholas da...@debsinc.com
snip
[callbob]
Exten = _XX.,1,answer
Exten = _XX.,n,Set(CALLERID(num)=${EXTEN})
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)
From memory, I did it this way :
Exten = _XX.,1,Set(CALLERID(num)=whatever)
Exten = _XX.,n,answer
Exten =
Hi Sammy,
Sorry for the previous answer. I accidentally pressed the send button. So,
I'm re-sending this mail with
my additional information.
Thank you for the reply.
I'm not using MixMonitor or Monitor. I'm recording the file in the perl
script that I call in my dial plan.
When there is an
Hello again,
Hmmm...So you are in the AGI, I'm not much into AGI stuff, but let me
reproduce this in my local env...BTW which asterisk version you are using !
--
Regards,
Sammy
On Fri, Nov 4, 2011 at 1:43 PM, Yaprak Ayazoglu
yaprak.ayazo...@gmail.comwrote:
Hi Sammy,
Sorry for the previous
I'm using Asterisk 1.6.2.7
I'm working on Ubuntu 10.10
Thank you for the reply.
On Fri, Nov 4, 2011 at 11:04 AM, Sammy Govind govoi...@gmail.com wrote:
Hello again,
Hmmm...So you are in the AGI, I'm not much into AGI stuff, but let me
reproduce this in my local env...BTW which asterisk
Hi Jean,
I suggest Queuemetrics. There are many out there but this one is good for
monitoring and reporting.
I know there's a free version you can try.
All the best
Anthony
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Can anyone tell me why I get this error very often?
[Nov 4 12:12:42] ERROR[21533]: app_queue.c:3083 ring_entry: Found a
channel matching iterface SIP/213_323 while status was 1 changed to 0
Regards
Thorben G. Jensen
--
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Hi,
does anybody have Augeas lens for Asterisk?
I was googling for a while but could not find any information.
http://augeas.net/
Thanks,
Vaclav Strachon
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Thank you guys for your response,
One FXS port can only handle one call. A PRI T1 gateway can handle 23 call
channels. A single T1 Data line with SIP can handle about 18 call
channels running G711, 37 channels running g729
I just want to make sure that a T1 Gateway (capable of 23 call
On Thu, 2011-11-03 at 18:50 +0530, amit anand wrote:
Hi you can use Absoulte timeout to set the time limit feature for the channel
Hi,
Thanks for the suggestion. It's good to know that absolute timeout
exists (I'd not noticed that before). However, it won't help here
because we're setting up
Hi,
DAHDI and LIBPRI are the standard versions from the Asterisk web site.
The Asterisk server has a Sangoma E1 card in it and the line is regular
ISDN30.
Cheers,
Kingsley.
On Thu, 2011-11-03 at 08:14 -0500, Danny Nicholas wrote:
Please elaborate on your flavor of DAHDI and LIBPRI and what
Hi,
Thanks. We've tried this and the first test call we did correctly
limited itself to 1755 seconds, so it looks like this is a workaround we
could use. We'll do a few more tests before assuming it's going to be
consistent but for now it looks encouraging.
Thanks once again.
Cheers,
Kingsley.
I realized there was an error in my last post. I meant analog gateway
plugged into and FXO port.
DIDs must start somwhere. And I am under the impression that the
telcos are the one that have
control over that? Therefore, we would first need an analog gateway
plugged into an FXO, before
being able
A telco could either give you a analog line like the old phone line which you
have at home with 1 number and 1 line or a T1 which comes from the telcos
office to yours and plugs directly into a digital gateway with 23 lines and
lots of numbers. and no need at all for analog gateways on the way
When you say 2/3 of calls, is there an inconsistency to the same recipient
or could it be a carrier issue (Verizon only, T-Mobile only, etc.)?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, November 04, 2011
What is your target PBX is it Asterisk?
If so your best method is to take calls in direct via SIP trunks, but there
are PRI and FXO options available as well. You can not use an FXS gatway to
plug to the Telco Service lines.
SIP Trunk - Asterisk or Like VOIP compliant PBX..
If your PBX is
It might be a good idea for you to describe your application and ask for
suggestions.
How many concurrent calls do you need to handle? Do you need a few (or many)
DIDs (actual phone numbers)? Are the DIDs in a single geographic area, or
scattered all over the country(ies)? Is your application
Hello Bryant,
I just realized how much information Nick has left out. Basically we
would like to function as a DID vendor.
Yes, everything on our end will be converted into SIP using G711 codec
. We have an OC48 coming into
our network, and a contact with the local telco here willing to supply
us
Why not go direct to Verizon Business (they provide nationwide wholesale SIP
services) or Level3 for your SIP interconnect? Leave the local telco out of it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hello Eric,
That is also a good idea. I am new to the VoIP world an do not know
who the major players are however,
will catch on really quick as my background is enhanced neuro
networks. I understand all the theory
behind compressions, codecs etc... Just trying to apply it in the real
world. That
This is only true for PRI, T-1 and other PSTN services.
The wholesalers take care of all everything to do with the PSTN side and number
ports, etc. Also check out Gafachi and Vitelity for service on a smaller
scale. Level3 and Verizon have some hefty mins/month commitments.
-Original
Berry
The local Telco's have control over the local phone numbers but they make
share/collocation/LNP agreements with other carriers and VOIP interconnect
carriers so numbers get swapped/leased and rented between different vendors. As
a VOIP interconnected carrier this allows us access to 90%
Hello Bryant,
Thank you so much for your insight. This is the exactly direction we are headed.
Collocating equipment to different regions here in Canada, and
performing least-cost
routing.
Thanks Again,
Nick.
On Fri, Nov 4, 2011 at 10:57 AM, Bryant Zimmerman brya...@zktech.com wrote:
Berry
2011/11/4 Danny Nicholas da...@debsinc.com
When you say 2/3 of calls, is there an inconsistency to the same recipient
or could it be a carrier issue (Verizon only, T-Mobile only, etc.)?
When we tested, calls were originated by 32 different cellphones to a
unique ISDN-connected Asterisk
I would recommended FOP2
Its awesome.
On Fri, Nov 4, 2011 at 3:04 PM, Anthony Laudini alaudini.lo...@gmail.comwrote:
Hi Jean,
I suggest Queuemetrics. There are many out there but this one is good for
monitoring and reporting.
I know there's a free version you can try.
All the best
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