Please note that I don't claim myself a guru, just happened to be working with
Asterisk for some good number of years, so probably know some stuff better than
others.
As for the number of lines, 1800 lines will come down to 1000 lines using AEL
but not the opposite.
When I'll be back home,
Hi,
I have the following problem. I have an xlite client registered with
asterisk server. If i dial say 1500 an FAGI script is invoked which plays a
greeting message. I now want to hold this call and play music on hold from
FAGI. How do i achieve this? Please suggest me.
Thanks in Advance,
Hi Jahnavi,
try StartMusicOnHold and StopMusicOnHold
On Wed, Aug 4, 2010 at 12:45 AM, Janu Mukherjee janu.mu...@gmail.comwrote:
Hi,
I have the following problem. I have an xlite client registered with
asterisk server. If i dial say 1500 an FAGI script is invoked which plays a
greeting
Hi,
I have an xlite registered with asterisk server. When i dial a number AGI is
invoked. and in this we are running to threads one to record files and one
to play files. So i dialed the extension and i started recording and playing
at the same time. On the xlite i m getting an indication when
On Wed, Aug 4, 2010 at 12:12 PM, Janu Mukherjee janu.mu...@gmail.com
wrote:
Hi,
Hi, please learn to ask questions.
I have an xlite registered with asterisk server. When i dial a number AGI
is
invoked. and in this we are running *to threads one to record files and
one
to play files.*
What
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
been able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
register = serverA:serverapassw...@ip_of_serverb_nic/serverB
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
een able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
egister = serverA:serverapassw...@ip_of_serverb_nic/serverB
Tilghman, thank you for your reply.
The mapping in RFC 3398 is logically correct therefore I do not need to submit
a suggestion to its editor.
The mapping in Asterisk 1.4.24 is the problem:
402 Payment Required is mapped to 16 Normal termination instead of 21 Call
Rejected.
Could you direct me
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten = s,1,Answer
Exten =
Hi Dan,
I had tried the new version of webmeetme i.e., 4.0.2
The recording works very well.
I see following php errors whenever i try to add in conference.
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: order in
Thanks Oliver.
I tried those approaches but they did not work.
However, I just found a workaround finally. The SIPAddHeader and SIP_HEADER
functions enabled me to get the callerid working.
Thanks again!!
From: asterisk-users-boun...@lists.digium.com
Hello,
In my Asterisk server when i try to set the value for the queue option Skip
Busy Agents in Freepbx GUI it is not being written into the backend file
queues_additional.conf. As a result sometimes agents in queue gets calls
when they are already busy with another call. So i set ringinuse=no
Hi!
Let's say I call by SIP/trunk1/number and the proxy server is
down, is there a way to getCHANUNAVAIL?
*CLI core show application Dial
Unfortunatelythe timeout parameter will not do the job for me. I need
somethingequivalentto qualify to monitor the outboundproxy.
Why not qualify and
The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required
is mapped to 16 Normal termination instead of 21 Call Rejected.
Could you direct me to the relevant file of code where these mappings
are done? Before reporting a bug I would like to confirm whether this
issue has been
Hi all,
I have problem when i transfer call from one queue extension to other queue
extension.
*Scenario
*some one call to DID 8833383932 which is assigned to queue1 and pickedup
by extension1 of queue1, Now extension1 transfer call to queue2's
exntesion2, extension2 picked up the call but no
Hello ,
I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My
current values are
AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx
25-30 % of all answering machines.
Anybody have any suggestion to improve the accuracy of AMD.
Thanks
--
Dear list,
I'm trying to get Asterisk to work dual-stack on Linux and I'm left with
a question.
Imagine that a user (on the road) connects to Asterisk from various
places. Many of them probably don't have IPv6 support yet. However, his
house and office do have IPv6 connectivity. I would like
Hi,
the basic settings are pretty good ones. What I did to do improve the
performance and prevent the false positives, I started to recorded every
call, and analyzed every incorrect detection :) Fairly soon I came with
optimal set for my environment:
initial_silence= 2500
greeting
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Janu Mukherjee
Subject: [asterisk-users] How to record a file and play some other file
atthe same time
Hi,
I have an xlite registered with asterisk server. When i dial a number AGI
is
Hi Aurimas,
Thanks for your thoughts on this. Can you please let me know how playing a
silent audio file before AMD will help to tweak the parameter values.
On Wed, Aug 4, 2010 at 8:30 PM, Aurimas Skirgaila a.skirga...@gmail.comwrote:
Hi,
the basic settings are pretty good ones. What I did
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino
Subject: Re: [asterisk-users] Tweaking AMD in Asterisk
Hi Aurimas,
Thanks for your thoughts on this. Can you please let me know how playing a
silent audio file before AMD will
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Regards
Alejandro
--
_
-- Bandwidth and
Alejandro Cabrera Obed wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Regards
Alejandro
Either RAID1 with a couple of spare drives or RAID5
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Subject: [asterisk-users] Asterisk and RAID
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Not really an asterisk
On Wed, 4 Aug 2010, Alejandro Cabrera Obed wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
RAID-10
If your controller supports it. If not, do it with
Steve,
Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming up
very dry. OpenBTS sounds like it would work, but is way too expensive to
roll out to residential homes.
On Mon, Aug 2, 2010 at 6:53 PM, Steve Kennedy steve-aster...@gbnet.netwrote:
On Mon, Aug 02, 2010 at
Manmohan wrote:
I had tried the new version of webmeetme i.e., 4.0.2
The recording works very well.
Great!
I see following php errors whenever i try to add in conference.
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: order in
Thanks Danny, What should be the length of audio file ?
On Wed, Aug 4, 2010 at 9:21 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino
*Subject:* Re: [asterisk-users] Tweaking AMD
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino
Subject: Re: [asterisk-users] Tweaking AMD in Asterisk
Thanks Danny, What should be the length of audio file ?
I'm supposing that 3 to 5 seconds should be ok.
--
in my case it's 0.1 second and I can confirm, that on SIP channels it really
helps.
On Wed, Aug 4, 2010 at 8:51 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino
*Subject:* Re:
On Wed, Aug 04, 2010 at 01:13:56PM -0400, Matt wrote:
Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming
up very dry. OpenBTS sounds like it would work, but is way too
expensive to roll out to residential homes.
Pretty much all Femtocells use 3G locally and send
Ok, here's the challenge:
I would like to be able to find, match - and then react - upon prompts
that are presented by the outbound/remote side of a call. Think mobile
phone and This user is temporarily unavailable.
Collecting a limited number of known prompt snippets should not be a
problem,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Subject: [asterisk-users] Identify remote prompts: Partial audio matching?
Ok, here's the challenge:
I would like to be able to find, match - and then react - upon
You might be able to record these snippets then pass them through the
Vestec or Lumenvox Speech engine to get what you want.
Unfortunately that won't work because:
* the containing recordings/feeds can be quite long, can be
embedded/surrounded by silence, ringing tones, music or special
Ot
Nuance for linux?
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Philipp von Klitzing
Skickat: den 4 augusti 2010 22:29
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re:
I have done an OpenBTS research and try project and OpenBTS is working
great. A complete set to roll out OpenBTS is not cheap but as far as I
know all femtocell kind of solutions need serious investments and
OpenBTS seems to be the cheapest among them. Asterisk is actually one
of the lego
Wednesday, August 4, 2010, 6:02:45 PM, Danny wrote:
R5 would use 3 out of 4.
You can have R5 across 10 drives too. Yes, the writes will be slow,
but it possible.
--
Best regards,
Gergomailto:csi...@gmail.com
--
Hello.
I have been beating my head over this problem for about 6 hours now.
I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:
[ Context 'default' created by 'pbx_config' ]
's' =1. Wait(1)
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood sch...@gmail.com wrote:
Hello.
I have been beating my head over this problem for about 6 hours now.
I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:
[
I don't see any
On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote:
You don't have any extensions in your default context that match the
extension that your sip peer is dialing in on. 's' is not a default
extension for SIP...try using _X., and see what you get. Bump up
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:
I don't see any
On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com
wrote:
You don't have any extensions in your default context that match the
extension that your sip peer is dialing in on. 's' is not a
Thx Rudi. but this query results in *Empty set (0.32 sec)
src AND dst like number *seems to be the problem area.
*
*
Also, how can I get the hold time talk time as separate values OR may be
total call connect time talk time (the difference of the 2 will be hold
time).
Thx
Sans
On Wed, Aug 4,
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:
I don't see any
On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com
wrote:
You don't have any extensions in your default context that
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood sch...@gmail.com wrote:
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com
wrote:
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:
My experience with Asterisk in the past has been with inbound analog
lines so that
Suddenly the other day we noticed MWI stopped working for SIP clients.
A sip show peer X returns this:
ast01*CLI sip show peer 719XXX
* Name : 719XXX
Realtime peer: Yes, cached
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context :
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