Thank to Frank and Steve for your answers
My understanding is that you need to place on operator premise an
equipment that checks first the availability of the user on VoIP. If
not registered, it's routing the call through the cellular network.
Is it correct ?
But during the handover (wifi to
On Sat, Oct 10, 2009 at 10:16:43AM +0200, Patrick wrote:
Thank to Frank and Steve for your answers
My understanding is that you need to place on operator premise an
equipment that checks first the availability of the user on VoIP. If
not registered, it's routing the call through the cellular
You can try free version of MOR Softswitch with billing and routing:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/
We rewrote Asterisk CDR completely and yes, it supports transfers.
More info about MOR: http://www.voip-info.org/wiki/view/MOR
Free version supports up to 10
can i use MP3 files as an IVR prompts directly without converting to .gsm
format? ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
Marco Mouta wrote:
Sad to say, but I believe this is only the small beginning….
Just a guess, and off-topic, but probably someone got very angry at
citibank. At least in Spain, they (or a marketing contractor) seem to
have called every single mobile phone in this country, they called me
five
I'm using Zap, not chan_local
i've tried to record the call and have seen that the audio DTMF toned
received is very poor, i've tried to put relaxdtmf=yes in zapata.conf
and increare rxgain and txgain from 0 to 5 but it doesn't seems to be
much better.
Is there something else to do?
Thanks
On
Hello all,
Is there anyway that I can configure Asterisk to start dialing out from fxo
after (xx) seconds from getting the dial tone? I don't want tdm card to send
the number immediately because it fails many times.
Thanks for any help.
___
--
The approach to Wi-Fi to GSM handover differs depending if its handled by
the cellular carrier or enterprise PBX.
If it's handled by the cellular carrier (as in the case of UMA) there's the
advantage of not tying up additional trunk lines for incoming calls, but
unless the rest of the
Looking at my shiny new Google Wave account, I was wondering if anyone else
on this list is in the beta AND going to Astricon. Astricon seems like it
would be a good test of the kind of collaboration GW is trying for. In any
case, I'd love to try to do an Astricon wave so let me know if you're
B.Masoud @ SH wrote:
Hello all,
Is there anyway that I can configure Asterisk to start dialing out
from fxo after (xx) seconds from getting the dial tone? I don’t want
tdm card to send the number immediately because it fails many times.
You can use the w. This is from the wiki:
If you
On Sat, 10 Oct 2009, gergis.rasmy wrote:
can i use MP3 files as an IVR prompts directly without converting to
.gsm format?
You don't want to do this.
Asterisk will attempt to use prompts encoded with the same codec being
used for the channel. So, unless you have a channel that is using MP3,
I'm having hard times with paging intercom
Heres my dialplan
exten = 777,1,Goto(intercom,777,1)
[intercom]
exten = 777,1,SIPAddHeader(Call-Info: sip:192.168.16.105\;answer-after=0)
exten = 777,2,Page(Local/3...@page Local/3...@page Local/3...@page)
[page] ; Paging
I use elastix,
I have this for dialout:
exten = s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
also what If I want 1 second delay?
thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
B.Masoud @ SH wrote:
I use elastix,
I have this for dialout:
exten = s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
right before the dialed number
If I understand your code it should be:
exten = s,8,Dial(www${OUT_${ARG1}}/${ARG2:${length}})
Remember that,
John Novack wrote:
B.Masoud @ SH wrote:
I use elastix,
I have this for dialout:
exten = s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
right before the dialed number
If I understand your code it should be:
exten =
Do you mean that incoming calls on your PSTN line works as
they should,
but not when they reach the voicemail? or that incomming
calls on PSTN
are always mute?
Incoming calls on PSTN line work as they should but, when someone leaves a
voicemail message the messege is mute. When I try to
Ivan Stepaniuk wrote:
John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
seems to contain the channel technology, the 'w' should be inserted
after the slash.
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I agree.
Doug
--
Ben Franklin quote:
Those who
On my Grandstream GXP 2010 I have the possibility for 6 channels and
thus 6 different accounts...
Line 1 I define an account that registers directly to an online
Asterisk-server, somewhere in a datacentre.
Line 2 I define an account that registers to the local Asterisk-server
(NSLU2 unslung)
On Sat, 10 Oct 2009, jonas kellens wrote:
On my Grandstream GXP 2010 I have the possibility for 6 channels and
thus 6 different accounts...
Line 1 I define an account that registers directly to an online
Asterisk-server, somewhere in a datacentre.
Line 2 I define an account that registers
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
Thanks.
-Original
B.Masoud @ SH wrote:
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
You need to type reload at the Asterisk console
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor
On Sat, Oct 10, 2009 at 7:59 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 10 Oct 2009, gergis.rasmy wrote:
can i use MP3 files as an IVR prompts directly without converting to
.gsm format?
You don't want to do this.
Asterisk will attempt to use prompts encoded with the same
On Fri, 9 Oct 2009, Ken D'Ambrosio wrote:
Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I
get:
Rejected connect attempt from 64.2.142.19, who was trying to reach
'6031234567@'
This leads me to my first question -- why doesn't it show a context? (My
second is,
Landy Landy wrote:
Do you mean that incoming calls on your PSTN line works as
they should,
but not when they reach the voicemail? or that incomming
calls on PSTN
are always mute?
Incoming calls on PSTN line work as they should but, when someone leaves a
voicemail message the
B.Masoud @ SH wrote:
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Is it now on work? Or I have to
On Sat, 10 Oct 2009, RSCL Mumbai wrote:
How should I convert my .wav prompts into aLaw, uLaw, G729 ?
The standard Asterisk prompts are already available in a wide variety of
encodings.
Try googling for asterisk convert mp3 to wav
Some will suggest to use Asterisk. Besides appearing to use a
Sorry for keep asking, but I did extensions reload, and restarted asterisk,
What should the message looks like? I still get the same:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Thanks for your help.
B.Masoud @ SH wrote:
Sorry for keep asking, but I did extensions reload, and restarted asterisk,
What should the message looks like? I still get the same:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Hi all,
After running for months without issue I've got a situation where
incoming SIP calls to my asterisk server are failing while outbound
calls appear to be working as expected.
The server is a gateway between my home LAN and a broadband cable
connection with a dynamic IP. The gateway runs
Asterisk to Asterisk voicemail not working (accessing voicemail from another
asterisk).
PSTN to Asterisk is working, but not between two asterisk :-(
I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto
--
Joseph
I'm trying create a feature that allows a callers to add more speech to his
recording. I think this can be done inside a dialplan, but I can't find an
example of how to do this.
Basically,after he records the primary message, a menu would play asking if he
wants to append to this message. If
exten = _X.,n,System(sox arg1 ... argN)
Martin
On Sat, Oct 10, 2009 at 5:25 PM, Bart Fisher b...@icpage.com wrote:
I'm trying create a feature that allows a callers to add more speech to his
recording. I think this can be done inside a dialplan, but I can't find an
example of how to do this.
If you are using freepbx, you have to change it in the GUI in the trunk
parameters.
B.Masoud @ SH i...@saudihome.com wrote:
Sorry for keep asking, but I did extensions reload, and restarted asterisk,
What should the message looks like? I still get the same:
-- Executing
Hardware echo usually helps. You can aslo try using OSLEC.
- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Friday, October 09, 2009 23:50
Subject: Re: [asterisk-users] choppy sound
Hi,
I am using CentOS
The record app has an append feature, if I remember correctly.
Bart Fisher b...@icpage.com wrote:
I'm trying create a feature that allows a callers to add more speech to his
recording. I think this can be done inside a dialplan, but I can't find an
example of how to do this.
On Sat, 10 Oct 2009, Bart Fisher wrote:
I'm trying create a feature that allows a callers to add more speech to
his recording. I think this can be done inside a dialplan, but I can't
find an example of how to do this.
Basically,after he records the primary message, a menu would play asking
Joseph wrote:
I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto
If understand correctly, you have two asterisk servers and when you dial
from one the other, DTMF is not recognized. I also asume you are using
SIP to connect
listm...@websage.ca wrote:
After running for months without issue I've got a situation where
incoming SIP calls to my asterisk server are failing while outbound
calls appear to be working as expected.
The server is a gateway between my home LAN and a broadband cable
connection with a dynamic
Steve Edwards wrote:
On Sat, 10 Oct 2009, Bart Fisher wrote:
I'm trying create a feature that allows a callers to add more speech to
his recording. I think this can be done inside a dialplan, but I can't
find an example of how to do this.
Basically,after he records the primary message,
On Sun, 11 Oct 2009 01:41:36 +0200
Ivan Stepaniuk i...@albafotonica.com wrote:
listm...@websage.ca wrote:
After running for months without issue I've got a situation where
incoming SIP calls to my asterisk server are failing while outbound
calls appear to be working as expected.
The
listm...@websage.ca wrote:
On the LAN side I can see the INVITE and OKAY messages which end with a
CANCEL, apparently initiated by the Asterisk gateway.
On the WAN side I can see that my Asterisk gateway is repeatedly
sending OKAY messages in response to the INVITE from my ITSP. I assume
Ivan Stepaniuk wrote:
listm...@websage.ca wrote:
On the LAN side I can see the INVITE and OKAY messages which end with a
CANCEL, apparently initiated by the Asterisk gateway.
On the WAN side I can see that my Asterisk gateway is repeatedly
sending OKAY messages in response to the INVITE
On 10/11/09 01:27, Ivan Stepaniuk wrote:
Joseph wrote:
I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto
If understand correctly, you have two asterisk servers and when you dial
from one the other, DTMF is not recognized. I
On Sun, 11 Oct 2009 02:11:47 +0200
Ivan Stepaniuk i...@albafotonica.com wrote:
listm...@websage.ca wrote:
On the LAN side I can see the INVITE and OKAY messages which end
with a CANCEL, apparently initiated by the Asterisk gateway.
On the WAN side I can see that my Asterisk gateway is
Steve Edwards wrote:
The dialplan snippet would look something like:
exten = s,n,system(/path-to-sox/sox a.wav b.wav c.wav)
This would copy a.wav followed by b.wav to a new file, c.wav.
On Sat, 10 Oct 2009, Barton Fisher wrote:
hmm, no luck. Here's what I have:
exten =
On Sat, Oct 10, 2009 at 11:47 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Sat, 10 Oct 2009, RSCL Mumbai wrote:
How should I convert my .wav prompts into aLaw, uLaw, G729 ?
The standard Asterisk prompts are already available in a wide variety of
encodings.
Try googling for
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