will be greatly appreciated.
- Bharath
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();
$self-agi-exec(MixMonitor, $ConversationFile|ba);
Any suggestions would be greatly appreciated.
Thanks a lot
Bharath
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Hi,
We are trying to implement a complex business logic in Asterisk. Executing
Wait_For_Digit command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?
Thanks
;
}
Is this not the correct way to do this? Or Are there any other methods?
Thanks
Bharath B. Reddy Bynagari
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Hi,
I am trying to implement a macro-screen mentioned at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I put the following code in my extensions_additional.conf
screen-from: You have a call;
screen-accept: Press 1 to accept this call or any other key to reject.;
);
---
The DIALSTATUS is empty.
But the Channel status is coming as 6.
I have the user decline the call and still not getting any DIALSTATUS
Thanks a lot in advance.
Thanks
MavenSphere-2-logo-web-orange
Bharath B. Reddy Bynagari
President CEO
Is there an option to specify a stun server in those IP phones?
On 12/19/05, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings all,A couple of clients have recently decided they'd like extensions to theiroffice PBXs at their homes, so they've duly been provided with preconfiguredphones which
You need to recompile asterisk after you install speex.
On 12/12/05, Steven [EMAIL PROTECTED] wrote:
I do not think that speex is installed by default.run show translations in asterisk and see what you get.StevenMay you have the peace and freedom that come from abandoning all hope of
having a
Forward UDP Ports 1-2 to your asterisk box.
On 12/8/05, Jeffery Chen [EMAIL PROTECTED] wrote:
can u paste your sip.conf general section,,?
there have another possible cause... the both side use different codecm and asterisk can not translaste it ...
-- Jeffery
On 12/8/05, chawki hammoud
I had the same problem, had to reboot the machine it started to work.
ThanksOn 12/3/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,I setup music on hold as directed for Asterisk 1.2 but still no music onhold. Any ideas what I did wrong. I see it start in the CLI but then itimmediately stops?? I also
I keep getting this error message from one of my Avaya 4620SW hard phone.
Got SIP response 400 Invalid Subscription-State back from
192.168.xx.xx which is the IP address assigned to that hard phone. Also
the phone will still have dial tone but cannot make or recieve any
calls.
Thanks
Hello All,
I'm using an Avaya 4620SW with Asterisk, the phone when hooked up to
the network, works for sometime, (I have not actually monitored the
time) maybe 20-30 minutes, after which the phone will still have a dial
tone, but can't dial out or recieve calls. I scanned thru the logs and
found
I had the UDP ports forwarded, In any case I will be testing with a
brand new router today, then will confirm if my old router had a
problem.On 11/24/05, Tom Rymes [EMAIL PROTECTED] wrote:
On Nov 24, 2005, at 12:14 PM, Bharath wrote: I found out that I have a faulty Belkin Router which
have you added allow=speex allow = ilbc in the sip iax conf files ?
On 11/25/05, Alejandro Vargas [EMAIL PROTECTED] wrote:
I'm testing [EMAIL PROTECTED] 2.0 beta 6.I'm checking de different codecs but with speex and ilbc I don'treceive any sound. I tested xtensofphone and iaxComm. With both I
I found out that I have a faulty Belkin Router which was causing the
problem. I tried forwarding ports as well as DMZ'd the Sip device but
still could'nt not hear the voice. So i plugged the sip device directly
to the cable modem it worked fine. So my guess is that though I
have set up the router
to do with codecs?
ThanksOn 11/22/05, C F [EMAIL PROTECTED] wrote:
On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution.
My Asterisk is on a public domain, there is no NAT
it work just by opening ports,
let me know..I have never been able to get it to work, that's why I don't
use sip, just plain iax2 for everything… J
Manny
-Original Message-
From:
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]]
On Behalf Of Bharath
Sent: Wednesday,
November 23, 2005 10:08 AM
Hello All,
I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution.
My Asterisk is on a public domain, there is no NAT or firewall in front
of the asteris box. I have sucessfully connected iax2 softphones
was able to recieve make calls. In the same
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