We have a lot of clients who run small call centers based on Trixbox, and
seem to be pretty happy with them. Have a look here:
http://queuemetrics.com/manuals/QM_Trixbox-chunked/
Thanks
l.
2010/3/31 Frank Church voi...@googlemail.com
On 29 March 2010 21:46, Frank Church voi...@googlemail.com
Hi,
the system() part pointed me in the right direction.. Thanks, going to give
it a test now..
Thanks!
Andy
On 29 March 2010 20:24, Zeeshan Zakaria zisha...@gmail.com wrote:
Hi,
I have done it a few times. Just posted a small blog about it with code.
Check it at
Jonathan Addleman j...@redowl.ca wrote:
nik600 wrote:
I was trying to record a call usng Mixmonitor and then convert it
using ffmpeg but the recording file is continuosly growing and ffmpeg
ends the conversion before of the call completion.
Here's my quick and easy eagi script:
I think you have caller ID update set to Yes and A2Billing first asks you
to: Enter your Caller ID number and then it asks you: Enter your
destination number while you mistake both for destination number.
Otherwise, I am confused by the title of your question that your caller id
doesn't pass and
SugarCRM and the church. This sounds just like a business; one that doesn't
like to call itself a business but employees tactics. I suggest providing
them with a solid cisco system with 100s of thousands dollars in cost where
they will have less money left to do bad things to world. Asterisk is
huu giang schrieb:
Hi Zeeshan
I know a solution using DRBD, Heartbeat and RedFone hardware to
provide failover ability to Asterisk.
If I have two Asterisk Servers, and each server has a TDM card and a
PRI line connect to each card, how your solution can provide failover
ability to
Hi,
Good to know this but I am not the poster of this question and not doing any
load balancing.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-03-31 7:33 AM, Tobias Wolf tobias.w...@evision.de wrote:
huu giang schrieb:
Hi Zeeshan
I know a solution using DRBD,
Many thanks Jonathan!
On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote:
What is the significance of /dev/fd/3 where does it come from?
I'ts the file descriptor 3 for the EAGI process, wich contains the audio.
--
/*/
nik600
http://www.kumbe.it
--
Hello.
I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a
bad quality of voice for incoming SIP calls into the app_meetme. As I know,
in my case of calls, jitter buffer is NOT executed on anyone channel. So,
after reading Russell Bryant's post (
OK, I see, but what I would really like to do is the opposite -- stream
an internet stream into a call or a meetme conference -- what would be
the best way on how to do that?
nik600 nik...@gmail.com wrote:
Many thanks Jonathan!
On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote:
On Wed, Mar 31, 2010 at 2:17 PM, cov...@ccs.covici.com wrote:
OK, I see, but what I would really like to do is the opposite -- stream
an internet stream into a call or a meetme conference -- what would be
the best way on how to do that?
And (hijacking thread with related question) I'd like to
Hello,
Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE
We have a very simple setup, using SIP softphones and a simple diaplan
as follows in the examples below. When I dial the 700 extension it
asks me for the extension and password, and it always says login
incorrect. The mail system send the email ok
The message Couldn't read user name means it is not receiving the DTMF. Do
you have an IVR to verify that your system is receiving the DTMF? If not,
setup one, call into it and send Dtmf to it and see if it responds at all.
If it doesn't, somewhere DTMF settings need to be adjusted.
Zeeshan A
I use this all the time and am very pleased with the results...
sox ORIG_FILE.WAV -r 8000 -c 1 OUTPUT_FILE.gsm resample -ql
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday,
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
The message Couldn't read user name means it is not receiving the DTMF. Do
you have an IVR to verify that your system is receiving the DTMF? If not,
setup one, call into it and send Dtmf to it and see if it responds at
On Tue, Mar 30, 2010 at 4:16 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello All
do you have ant software in order to change the format from mp3 or wav to
gsm in order to using it in asterisk file
thank you so much for your help and support
Best Regards,
salah
If you
Another option is to tie in a legacy 2-wire PBX with Asterisk instead of going
pure analog
This allows you to reuse your single-pair infrastructure, while achieving MOST
of the functionality of a pure-ip endpoint deployment with only a very moderate
incremental cost over a pure-analog
A (hopefully) helpful addition to this reply; ORIG_FILE.WAV is a wav49 file.
If the name is orig_file.wav, it is a regular wav file and the sox command
would generate (IMO) a better output like this:
sox orig_file.wav.WAV -r 8000 -v 10 -c 1 OUTPUT_FILE.gsm resample -ql
_
From:
Hi,
Oki, thank you so much for this solution i really appreciate it
Regards,
Salah
2010/3/31 Danny Nicholas da...@debsinc.com
A (hopefully) helpful addition to this reply; ORIG_FILE.WAV is a wav49
file. If the name is orig_file.wav, it is a regular wav file and the sox
command would
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of JR Richardson
Sent: Tuesday, March 30, 2010 6:55 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dropped Calls
I've written about
Hi list,
can anyone tell me how to reset/delete all modifications (personal
greeting message, personal name, ...) I made in my voicemail?
I just want to get the default automatic computer messages back.
thank you!
greets
felix
--
On 3/31/2010 10:38 AM, Michael L. Young wrote:
Is there a chance that you are using Realtime at all?
I am just curious because I was having problems with dropped calls as well
and just discovered that it appears to be related to the database server.
If for some reason on the database server
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
The message Couldn't read user name means it is not receiving the DTMF. Do
you have an IVR to verify that your system is receiving the DTMF? If not,
setup one, call into it and send Dtmf to it and see if it responds at
Hi!
And (hijacking thread with related question) I'd like to stream from
an incoming leg of a SIP channel to the Internet. Any suggestions on
that?
Start here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ices
Philipp
--
On Wed, Mar 31, 2010 at 6:27 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Start here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ices
Thanks, Philipp
--
_
-- Bandwidth and Colocation Provided
Just to get a 100% correct response to last question, are you using the flat
CDR or mysql/some other DB?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson
Sent: Wednesday, March 31, 2010 10:58 AM
Hi!
I am just curious because I was having problems with dropped calls as
well
Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
All extensions are hard-coded. We only have a handful of
phones that don't change.
This last sentence is a wounderful example of a
Hi,
Vinicius, did you actually solve the choppy audio issue by compiling
Gordon's kernel? I have the same problem on the exact same Alix platform
(using kernel 2.6.31, though).
regards,
Darko
ps. Sorry everyone if this mail does not get threaded right. I've just
joined the mailing list and not
Thanks Andrew,
I have some doubts regarding this issue, firstly, i'm using Asterisk 1.4.21
not 1.6 like the issue you showed to me (
https://issues.asterisk.org/view.php?id=16887) other thing is that i have
many other asterisk servers working good and i never made this change
By the way i'm
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from one
of my servers to the phones results in garbled audio. Anyone else been able to
make this work
Hi,
it was some configuration error. I droped the old config and started
with the sample file to build a new one. Therefore I do not know
which parameter in chan_dahdi.conf caused the problem. Anyway, now
it is working.
The only remaining problem is that no caller ID is available for
incoming
According to this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
, you could pipe the stream into the conference using an AGI script. I
haven't actually tried it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Jonathan C. Bailey wrote:
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from
one of my servers to the phones results in garbled audio. Anyone else
Hi!
I want to get audio from the PSTN before the call is answered so I don't miss
when the called phone is busy or if there is some error (like the phone is
unavailable or is wrong, etc) and hear the ringing from my telco.
I have polarity reversal in my telco for incoming and outgoing calls.
Maintenance of Asterisk 1.6.0 and 1.6.1 will move to security fixes only
in approximately one month. There are bug fix releases scheduled to be
released during the first half of May for both versions. After those
releases, Asterisk 1.6.0 and 1.6.1 will only receive security fixes.
The Asterisk
Hi,
We are using Asterisk and PERL. We have all the call logic in PERL. We are
trying to identify the caller using the CID in the Database. As the Database
lookup is taking more time (15 seconds), we want to play some tune while
the caller is waiting.
How can we do that? Any ideas will be
Bharath B. Reddy Bynagari wrote:
Hi,
We are using Asterisk and PERL. We have all the call logic in PERL. We
are trying to identify the caller using the CID in the Database. As the
Database lookup is taking more time (15 seconds), we want to play some
tune while the caller is
Hi!
I'm trying to play a pre-recorded message to a group of Aastra phones
[...]
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.
20,port-audio=16000,proto=udp}
Look at the very bottom of this (snom multicast):
On Wed, 31 Mar 2010, Bharath B. Reddy Bynagari wrote:
We are using Asterisk and PERL. We have all the call logic in PERL. We
are trying to identify the caller using the CID in the Database. As the
Database lookup is taking more time (15 seconds),
Fix the database! Anything else is a
Felix Tiefenthaler wrote:
Hi list,
can anyone tell me how to reset/delete all modifications (personal
greeting message, personal name, ...) I made in my voicemail?
I just want to get the default automatic computer messages back.
thank you!
greets
felix
If you are storing
I think I may have to do that.. I'm beginning to think my idea with VLC just
won't work. BTW, we're running 1.4.28 (but so far there seems to be a backport).
- Original Message -
From: Leif Madsen leif.mad...@asteriskdocs.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi there!
just a quick question: what would you recommend to get to connect an
asterisk box to the analog phoneline?
I have two linksys spa2102 and a sap9000 but as far as I know I need
something else to connect the asterisk box to the analog phoneline. I
just have two analog phone lines, so
On 3/31/2010 12:06 PM, Danny Nicholas wrote:
Just to get a 100% correct response to last question, are you using the flat
CDR or mysql/some other DB?
All sip clients/peers are defined in sip.conf, dial-plan is entirely in
extensions.ael. We have one office that uses an Asterisk native
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
I was suspecting something with either rtptimeout or sip registration
timeout, but I'm not sure what.
--
Do you mean that SS7 switch is a MSC and do all MSC support load balancing
without any hardware between it and my Server.
Sorry for my English, what do you mean two point codes for my servers ?. I have
at least two servers.
--- On Wed, 3/31/10, Tobias Wolf tobias.w...@evision.de wrote:
45 matches
Mail list logo
