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...@gmail.comwrote:
Hi Gopal,
Is there a way to Configure OutBound IVR. Correct me if i might have missed
reading the URL you pasted.
Thanks
Kaushal
On Wed, Apr 20, 2011 at 1:04 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Try this
http://www.freepbx.org/support/documentation/administration-guide
...@gmail.comwrote:
On Wed, Apr 20, 2011 at 2:56 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Outbound call only you can make like dial out rules, IVR is only for
Inbound calls. In outbound I dont how you are asking this IVR facility. You
mean like voice broadacasting? like dial the number
Hi,
Can anyone suggest which is the best scripting language for Asterisk or any
telecom device? Thanks in advance.
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Thanks for your suggestion:)
On Fri, Apr 1, 2011 at 5:30 PM, mahesh katta maheshka...@flexydial.comwrote:
Perl is the best script
On Fri, Apr 1, 2011 at 5:27 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
Can anyone suggest which is the best scripting language for Asterisk or
any
from my iPhone
On Apr 1, 2011, at 8:27 AM, Roger Burton West ro...@firedrake.org
wrote:
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can anyone suggest which is the best scripting language for
Asterisk or any
telecom device?
Depends on the other parameters
wrap it around the (excellent) asterisk-java framework and
have clean simple access to AMI and AGI interfaces.
Alternatively look at adhearsion - which is a ruby framework for asterisk.
But it _really_ does depend on what you are doing.
T.
On 1 Apr 2011, at 12:57, Gopalakrishnan A.N wrote
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Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id checkbox while creating VoIP trunk then it started
working for me..
On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Please try this in your dialplan Set(CALLERID(name
I guess it will not work with PSTN lines since the control is transferred to
the Exchange. I am not too sure, I am just sharing my thoughts
On Fri, Nov 19, 2010 at 9:28 PM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi Gopalakrishnan A.N,
I tried it but it seems like my
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Hi,
I am facing some voice drop in inbound, outbound, and IVR. But while
checking the process of the CPU and memory utilization is very less.
Mem: 21304K used, 36500K free, 0K shrd, 1896K buff, 13228K cached
The voice drop is in systemic. I am not too sure what to check... all the
configuration
Hi James,
I too facing the same issue whereas in the inbound call I am able to receive
the call, when I pickup the receiver it hangsup. I am getting the NOTIFY
option.. the log as follows,
-- SIP read from 98.158.181.173:5060:
NOTIFY sip:pbxfami...@10.0.8.84:5060 SIP/2.0
Via: SIP/2.0/UDP
/listinfo/asterisk-users
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this happens when there is no reply from the DNS. So change DNS or install
it locally on your asterisk server. At least caching name server should be
installed.
Zeeshan A Zakaria
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On 2010-09-24 1:51 PM, Gopalakrishnan A.N sai...@gmail.com wrote:
Still I have
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...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *ext Gopalakrishnan
A.N
*Sent:* Thursday, September 23, 2010 5:46 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Record() Cmd and My SQL
I hope it cant be done using
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Hi,
How to create dialplan restriction for particular extensions..
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as busy tone is detected because of this
existing call is disconnected.
Did anybody faced this kind of issue. Also some assistance would be much
appreciated.
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By somehow I made it work by having T38 passthru in both Asterisk and
SPA3102.
Thanks for the comments..
On Tue, Sep 14, 2010 at 7:05 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
I tried to send fax from Linksys to Grandstream by configuring openSER
account.. that works fineonly
Hi,
I tried to send fax from Linksys to Grandstream by configuring openSER
account.. that works fineonly when I send fax from Linksys to Asterisk I
am not able to send
On Thu, Sep 9, 2010 at 8:42 PM, Gopalakrishnan A.N sai...@gmail.com wrote:
I am from India and I hope I have
checking
in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas
from Asterisk to SPA3102 I can able to see some rtp traffic.
Your help would be much appreciated
On Tue, Sep 7, 2010 at 10:30 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
I have created one SIP extension
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Gopalakrishnan A.N sai...@gmail.com wrote:
Hi,
I have created one SIP extension in Asterisk and configured that
extension in SPA 3102. And connected one FAX machine to the SPA3102 and
one to Asterisk.
The problem is if I try
/listinfo/asterisk-users
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checking
in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas
from Asterisk to SPA3102 I can able to see some rtp traffic.
Your help would be much appreciated
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,
Gopalakrishnan A.N,
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I am logging into asterisk manager thru a Java program but not able to
login, if i use PHP I am able to login. I have attached my java code with
this mail. Can someone step me up to go ahead
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Echoclient.java
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