Displaying 20 results from an estimated 1000 matches similar to: "can anyone tell me how to set asterisk to record all phonecall"
2005 Aug 26
0
PhoneCALL version 1.0 Administrative Manual - Released
Greetings Everyone!
The version 1.0 of the PhoneCALL Administrative Manual has been released.
It is more of an outline of the features and interface, and we'll be
adding lots of more detailed information in the manual over the next few
days/weeks.
Of course, we'd love to get your input on the manual and areas we need
to clarify or even some new sections in the manual that would help
2005 Sep 12
1
Phonecall or something as robust
Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/
Feedback?
Is there something as good as it or better ?
Recommendations?
--
=================================
Joshua Abbott, Support Technician
http://www.successfulhosting.com/
Direct Line: PENDING
Phone: (866) 494-5096 x1207
E-Fax: (419) 858-3241
Alt E-Fax: (801) 217-1123
jabbott@SuccessfulHosting.com
2006 Jun 15
1
Need to Hire: PHP Programmer for PhoneCALL
Hello all!
It's come time where I need to add another programmer to our team.
You should have at least 3 years of "work" experience with PHP/MySQL.
Please send me your resume and a few code samples if you can.
If you can only work part-time or full-time, please include that in your
response.
Along with your salary requirements.
You'll be working with PhoneCALL, so be sure to
2005 Aug 16
2
PhoneCALL v2.6.1 - Released
Hello All!
Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version
2.6.1 has been released, and is the current stable release.
http://www.vecsector.com/phonecall
We're always looking for feedback/testers to help us enhance it and make
it even easier for everyone to use. The current version is designed
around the advanced Asterisk user, and we are working on a more
2003 Sep 06
2
[LLVMdev] languages, semantic trees, LLVM interfaces
Hello Vikram,
Saturday, September 6, 2003, 9:10:45 PM, you wrote:
VSA> For any language with relatively sophisticated syntax and semantic
VSA> rules, you will probably need a higher-level representation like an
VSA> Abstract Syntax Tree in order to do type-checking and other kinds of
VSA> checking.
OK, concerning AST -- I see. Thank you.
VSA> For OCAML, for example, the
2003 Sep 06
2
[LLVMdev] languages, semantic trees, LLVM interfaces
Hello LLVM fathers,
1. "languages, semantic trees"
what do you think ideally, do languages implementations based on
LLVM need internal semantic tree or they should rather try to use
LLVM directly in/after syntax parsing?
For languages like C++ the expected answer is "of course we need
an internal semantic tree between parsing and LLVM!"
But I am still
2007 Jun 25
7
R-excel
Good morning to everybody,
I have a problem : how can I import excel files in R???
thank you very much
Dr.sa. Erika Frigo
Università degli Studi di Milano
Facoltà di Medicina Veterinaria
Dipartimento di Scienze e Tecnologie Veterinarie per la Sicurezza Alimentare (VSA)
Via Grasselli, 7
20137 Milano
Tel. 02/50318515
Fax 02/50318501
[[alternative HTML version deleted]]
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift <thanks darren!>) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
i've captured with:
core set verbose 10
core set debug 10
fax set debug on
sip
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the
2012 Sep 20
6
accept email and make phone call?
Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
joe a.
2015 May 05
2
[LLVMdev] llvm DSA - reproduce the result in PLDI 07 paper
Dear John,
I intend to implement the improvements on DSA.
After running DSA on SPEC, I found DSA gives low precision for mcf and
bzip2.
I have checked the most imprecise c files in mcf an found that the code
seems to be a mixture of "PHI" and "GEP" instructions.
Could you please give me some hints about what the big picture of the
improvement should be and how to start?
Thank
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' -
No matching peer found
my logger.conf
2012 Jul 18
4
asterisk 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be moments of stutter - perhaps 1 second of stutter for every
10 seconds of Playback. The stutter is not consistent at the same point
of the playback file.
To
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug.
can anyone suggest how i can further deal with this?
--
Jeremy Kister
http://jeremy.kister.net./
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
2011 Jun 06
2
issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at
http://issues.asterisk.org
when I click on 'My View' or 'View Issues' I get an error:
APPLICATION ERROR #401
Database query failed. Error received from database was #1142: DELETE
command denied to user 'mantisreadonly'@'localhost' for table
'mantis_tokens_table' for the
2013 Sep 10
3
Asterisk 1.8 drop calls after 15 minutes
Hi all,
I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
through OpenVPN seems to have the problem.
From CDR, I see for 3 calls from this morning I'm aware of, that
asterisk hangup after respectively 899s 894s 898s
In logs I see
WARNING[8213] chan_sip.c: Retransmission timeout reached on
2013 Oct 10
1
asterisk 11.6 nat problem
using asterisk 11.6.0-rc1 i just converted my "nat=yes" to
"nat=auto_force_rport,auto_comedia"
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the problem.
i left 'sip set debug peer vgw1' on the console. but i dont see what's