Thanks for reply,
This recording is meetme conference recording. normally meetme file is can
wav format. is there any ways to change the meetme file into gsm format.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201,
hi
as you know meetme default recording file format is wav file. you may change
is too gsm for reduce file size.
or if you want then you may use monitor or mixmontor for gsm recording too.
On 6 Oct 2011 12:09, mahesh katta maheshka...@flexydial.com wrote:
Thanks for reply,
This recording is
Hi,
I would like to receive an AMI event whenever an hint value changes.
If I'm not mistaken, this is not possible at the moment (you must query
current hints status and parse response), right ?
Would you appreciate this feature ?
I would see it as cheap way to mimic BLF status on a desktop
Would ExtensionStatus provide the data your looking for?
Event: ExtensionStatus
Privilege: call,all
Exten: 216
Context: ext-local
Hint: SIP/216
Status: 0
On Thu, Oct 6, 2011 at 9:12 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I would like to receive an AMI event whenever an hint value changes.
hello,
when i use the number of the first provider like that
exten = 520870900,1,Answer
exten = 520870900,n,Wait(4)
exten = 520870900,n,Meetme
All works without problem,the issue just with the second provider i use just
the last 3 numbers for the outbound all works without issue, but whe i use
thanks guys, I think I might just tell the customer to suck it up and
use his transfer soft key. He is using # as defined in features.conf
On 11-10-05 03:24 PM, Kevin P. Fleming wrote:
On 10/05/2011 02:03 PM, Danny Nicholas wrote:
Depending on hardware and number of parking lots, could hints
Unfortunately I don't know behaviour of Progress() function, so cannot
make any conclusions. As far as I traced it back to tech-
implementation, this call does not changes any state of channel. But I
analysed only sip and dahdi drivers. Neither it plays any indication
tones.
2011/10/5 Sammy
Looks like you do not have chan_dahdi.so loaded in Asterisk.If you don't
install DAHDI before you install Asterisk, then Asterisk will not be built with
support for DAHDI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I would change DAHDI/1 to DAHDI/G1 or DAHDI/R1 - DAHDI/1 is DAHDI line 1
only, R1 is Round Robin Group 1, G1 is sequential Group 1
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 10:46 AM
To:
Hey,
How've you configured your Outbound trunk ? DAHDI/1/04712527270 : What do
you've in your dahdi configuration file ! I doubt this /1 is the culprit
or else your DAHDI channel is not really working at all.
Regards,
Gohar A.
From: asterisk-users-boun...@lists.digium.com
If DAHDI is not really configured or chan_dahdi isn't loaded the the error
mesage would be can not create channel of type DAHDI but here its not the
case. Dadhi module is indeed loaded but the DAHDI device is not working
properly.
On Thu, Oct 6, 2011 at 8:49 PM, Gohar Ahmed gohar.ah...@vopium.com
Hi,
astrisks*CLI* module unload chan_dahdi.so*
Unable to unload resource chan_dahdi.so
Command 'module unload chan_dahdi.so ' failed.
Producing some other error messages !
On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote:
In the Asterisk CLI run the commands module
What happens when you do the module load chan_dahdi.so command?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 12:10 PM
To: Asterisk Users Mailing List -
Hi,
I'm looking for an old-style receptionnist SIP phone with the requirements
bellow.
I've found one theorically matching these (Yealink plus 3 LCD expansion
module).
Would you recommend an other one ?
My requirements are :
- each LCD expansion module should display at least 30 extension
Hi, folks.
I'm having a heck of a time trying to get outgoing T38 faxing (I don't
need inbound right now) working with FFA and Gafachi. G711 faxing works
(as well as can be expected over the internet), but I want the higher
reliability of T38.
I'm running Asterisk 10-beta1.
When I drop my
I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
signaling. Has anyone had any experience with these devices? The
feature cards that Cisco sells can be a little confusing. I'm
thinking something like below is what I need.
(1) AS5400XM, AS5400XM Starter Kit (inc Chassis,
On Wed, Oct 5, 2011 at 2:04 PM, Kyle Sexton k...@mocker.org wrote:
Does anyone know if there is a resource to see what changes were made
between different versions of Digium cards? For example, how
different is a TE410P revision C when compared to a TE410P 5th
generation card?
I know there
On 10/6/11 11:25 PM, Kyle Sexton wrote:
I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
signaling. Has anyone had any experience with these devices? The
feature cards that Cisco sells can be a little confusing. I'm
thinking something like below is what I need.
(1)
Also used for calling card platforms :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas
Sikkema
Sent: Thursday, October 06, 2011 5:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set debug
on, rtp set debug on and udptl set debug on
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Fri, Oct 7, 2011 at 1:37 AM, James Sharp ja...@fivecats.org wrote:
Hi,
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