--
calendar3ewsfree
Please help me out for solve above problem.
Thanks in advance
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On Sun, May 6, 2012 at 5:14 PM, Michel Verbraak mic...@verbraak.org wrote:
On 30-04-12 11:09, Bharat Lalcheta wrote:
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical
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1001 1001 28
I can receive listen and also do all stuff using voicemailmain
application. But no MWI on any client.
is there any thing else i need to check ? can any one help to solve the problem
Thanks in advance,
Bharat Lalcheta
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Hi you can get some help using n-way dialplan example. Its generate new
call and transfer current call in conference meetme. You can google to find
its example
On Nov 15, 2012 8:15 PM, Michael voip.quest...@gmail.com wrote:
Hi Aldo,
Thank you very much for answering my question.
Can you
as user complete his talk, admin
again press some digit and that user get muted.
Regards,
Bharat Lalcheta
On Wed, Dec 19, 2012 at 7:41 PM, pankaj pandey pankaj.n...@yahoo.comwrote:
conference, when QA session begins, is there a way for participants to
raise hands, if they have any questions so
I dont think this is existed.
However, its easy to build a script in php or perl or any other language
which check time from file or database and generate call file which execute
paging in asterisk. Just put this script in cron. Thats it...
Regards,
Bharat Lalcheta
On Thu, Dec 27, 2012 at 1
Its maximum call Bit rate available for that peer. Default is 384 kbps.
Your call for that peer allowed max bit rate or bandwidth of 384 kpbs only
Regards,
Bharat Lalcheta
On Fri, Jan 4, 2013 at 7:09 PM, XBrian bobo...@yahoo.co.uk wrote:
Hi
sip show peer 21342
gives me peer 21342's
Please study meetme application's options. You will get almost all feature
you ask for in it
On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de wrote:
Hi list,
I am in need of a special asterisk conference room with the following
constraints:
- there is one admin / moderator and several normal
webinar every Thurs:
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again.
Hope it helps you out.
Regards,
Bharat Lalcheta
On Mon, Mar 4, 2013 at 5:57 PM, Gertjan Baarda gertjan.baa...@gmail.com wrote:
Dear guru's
Hopefully someone can shed some light in my issue. I have created a queue
with a ringall strategy and all works fine. I want a caller to be able
No its again place into queue so its start with new available position.
However, mostly all users remain in same position if he come again in
queue using below scenario.
Regards,
Bharat Lalcheta
On Mon, Mar 4, 2013 at 6:22 PM, Gertjan Baarda gertjan.baa...@gmail.com wrote:
Ah.. thanks
yes,
context parameter in queue.conf is more likely option for you. It will
work during MOH too.
On Mon, Mar 4, 2013 at 6:33 PM, Bharat Lalcheta
bharatlalch...@gmail.com wrote:
No its again place into queue so its start with new available position.
However, mostly all users remain in same
voicemail
and hangsup the call.
If you want to give option to user for select voicemail or come back
to queue, you can use time out option.
Regards,
Bharat Lalcheta
On Mon, Mar 4, 2013 at 6:57 PM, Gertjan Baarda gertjan.baa...@gmail.com wrote:
ok, resumé: When I use the n option in the queue
Can you provide OS details ? Its seems problem of abrt. Did u tested
asterisk without abrt
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
Hi,
I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed
Did u test it without abrt?
On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com wrote:
Its Centos 6
with kernel 2.6.32-279.19.1.el6.x86_64
Regards,
Zohair Raza
On Thu, Mar 7, 2013 at 8:28 AM, Bharat Lalcheta
bharatlalch...@gmail.comwrote:
Can you provide OS details
channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan
time in you billsec.
But you are getting confused on answered word, it does not mean answer by
agent.
If you want agent talk time with your customer, then i think CDR is not
provided the same.
Regards,
Bharat Lalcheta
On Mon, Mar 18, 2013 at 10:59 AM, RSCL Mumbai rscl.mum...@gmail.com wrote:
I
computer.
Any ideas on a next step for debugging? I was thinking I would start a
wireshark trace to see if the rtp packets are actually leaving the client
computer.
Mitch
On 03/19/2013 08:28 AM, Bharat Lalcheta wrote:
rtp set debug ip 1.2.3.4
where 1.2.3.4 is ip of your particular agent
Did u changed rtp.conf ?
port is showing 39408. Asterisk definetly drop rtp packet for this port if
not updated in rtp.conf
Regards,
Bharat Lalcheta
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What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and
If u want to dial in round robin use Dial(zap/r2/2) . It dials using
channel in round robin
On Mar 21, 2013 9:37 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question
Use r2 instead of g2 in dial
Dial(Zap/r2/${EXTEN}
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${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok
and agents.
Regards,
Bharat Lalcheta
On Fri, Mar 29, 2013 at 4:41 AM, Chad Wallace
cwall...@lodgingcompany.comwrote:
On Thu, 28 Mar 2013 14:55:45 -0500
Gregory Malsack gmals...@coastalacq.com wrote:
History ~
I recently took a position with a call center. At the time they had
about 50
Hi,
You can check extension status using chanisavail function. And extension is
not free, you can divert your call to queue.
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail
Regards,
Bharat Lalcheta
On Thu, Apr 11, 2013 at 1:38 AM, Tommy Cooper tomcoope...@yahoo.com wrote:
Hi
Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and
not able to generate this scenario.
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Backtrace and logs attached here :
https://issues.asterisk.org/jira
reasons), the
; information will not be removed from
realtime storage
Also remove all qualify related parameters and keepalive if set
Hope it will solve your problem
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza
engineerzuhairr
} =
${CONF_PIN}]?getpin,good_exten,1)
Hope it helps,
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.netwrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each
,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 5:36 PM, Pat Collins drdialt...@optonline.netwrote:
Thank you Bharat.
Sadly, that made no difference.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bharat Lalcheta
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