Displaying 20 results from an estimated 500 matches similar to: "Blind Call Transfer using Sipura 3000 + aste risk"
2004 Aug 11
1
Blind Call Transfer using Sipura 3000 + asterisk
Hi List,
I hope this setup must be done by our astersik users..
I am using Sipura 3000 to receive PSTN calls and forward those calls to
asterisk for voice processing and after that, I am transferring call to
extension through FXS port on SPA 3000.
Currently, media of call is trombone through asterisk. i.e achieving blind
transfers on asterisk with SPA 3000.
Is it possible to stop trombone
2004 Jul 12
1
SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
into some sort of a codec mismatch or something because it's not working
right. The SIP client is a SPA-3000.
In iax.conf, I have something like the following:
[General]
2005 Sep 06
2
Wireless router with built-in VOIP(FXS) ports for Ansterisk
Hi Team,
Just, I would like to know, is there any unlocked device(wireless router
with built in FXS port) for home users which are connected Asterisk based
VOIP service.
I have looked products from Linksys and D-link etc. But all these products
are bundled with VOIP Service providers (vonage, lindo and at&t) .
Please sugest if any products avaiable in market.
Thanks,
Karun
2004 Aug 24
0
Perl AGI - no output from agi script to Aste risk
print to standard error output in your perl script:
print STDERR "This is how perl-AGI prints to Asterisk CLI output\n";
MATT---
-----Original Message-----
From: Robert Rozman [mailto:rozman@fri.uni-lj.si]
Sent: Tuesday, August 24, 2004 8:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Perl AGI - no output from agi script to
Asterisk
Hi,
2003 Aug 06
0
(no subject)
A new recording is now up on pan.zipcon.net and also gas.zipcon.net. Here is
the Readme:
The Virtuoso Trombonist
Dennis Smith plays with W.W.S.S Wind Ensemble Wm Cole, Conductor, and
with Martha Goldstein, organ. At ogg q=7.
<p>In the years befor world war 2, Sunday afternoons might be spent in the
park listening to the Municipal band. Selections 1 and 2 would be this kind
of
2020 Aug 17
0
getent passwd blank response
On 17/08/2020 13:18, Robert E. Wooden via samba wrote:
> root at mbr04:~# wbinfo --group-info 'Domain Users' | awk -F ':' '{print
> $3}'
> 10000
> root at mbr04:~# wbinfo -i [username] | awk -F ':' '{print $3}'
> failed to call wbcGetpwnam: WBC_ERR_DOMAIN_NOT_FOUND
>
> Could not get info for user [username]
Hmm, I only get that if I
2004 Jul 14
8
spa-3000 review?
Since the 3000 has been out for a little while, has anyone done a
review of the product? (couldn't find anything on google for wiki).
Can the fxo and fxs ports be used as two independent channels?
Is it really read for prime time?
Etc.
Rich
2004 Jul 06
2
How do I disable '#' to transfer a call?
I don't see anything on the Wiki or in the documentation about disabling
this feature.
2017 Dec 07
0
Revolutions blog: November 2017 roundup
Since 2008, Microsoft (formerly Revolution Analytics) staff and guests have
written about R at the Revolutions blog (http://blog.revolutionanalytics.com)
and every month I post a summary of articles from the previous month of
particular interest to readers of r-help.
In case you missed them, here are some articles related to R from the
month of November:
R 3.4.3 "Kite Eating Tree" has
2010 Mar 08
0
Is it possible to configure Asterisk so that it does the Q.SIG Path Replacement Feature ?
Hello,
If I connect an Asterisk 1.6 to a PBX via Q.SIG and
A (on the PBX) calls B (a SIP phone on Asterisk).
B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer.
Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ?
The Q.SIG "Path Replacement Feature" requires the following:
After both legs of the
2004 Aug 10
1
SIP Transfers (Possibly reinvite)
Hey Folks,
Is it possible to transfer an incoming call back out without a "trombone"
effect.
For instance;
Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the
caller selects an option --> asterisk transfers the call to my cell phone
via broadvoice and removes itself from the equation so I end up with...
Caller --> Broadvoice --> Cell Phone
Vs.
1998 Jun 29
3
Samba/printing
Hi all. I'm writing because printing works.... just not very well. I have
the samba server running on a Solaris 2.5.1 600MP box.
Here's the symptons:
Everytime a samba client send print requests to the samba server/print
server the job does print but I get a header and trailer page enclosing
each job. This means everytime a samba client send a print request two
sheets of paper are being
2004 Aug 12
2
Interruptable SayUnixTime
I'd like to announce the time when people call and hit my voice-menu
context, but the complaint is that the time announcement isn't
interruptable. Is there any way to make SayUnixTime interruptable?
-- PhoneBoy
2020 Nov 01
6
analyzing results from Tuesday's US elections
Hello:
What can you tell me about plans to analyze data from this year's
general election, especially to detect possible fraud?
I might be able to help with such an effort. I have NOT done
much with election data, but I have developed tools for data analysis,
including web scraping, and included them in R packages available on the
Comprehensive R Archive Network (CRAN)
2001 Jun 28
0
Finale and Maestro fonts
I have been successful in installing and running Finale 2000 with Wine.
The only - major - problem is that the music symbols are replaced by
squares (like the default symbol caracter), making the music unreadable! I
thought that importing the Maestro fonts with DrakeFont (Mandrake 8.0) and
making them available to the system would help, but it doesn't. I have
tried to add :
Alias0
2001 Jul 27
0
Font / charset encoding problem?
I have successfully installed and launched Sibelius (music typesetting
software) on linux (with codeweavers preview 4, w/ a fake_windows install).
It works fairly well, and would probably deserve a 4 on the app database.
Except for the fonts, which kind of defies the purpose of a typesetting
software! Here are the symptoms and some log activities :
* No musical symbols are displayed (except
2007 Aug 29
0
Hangup detection and trombining
Hi All,
I hate to post yet another "bloody hangup detection issue" on the list, but
I have been pulling my hair out no end of late with a hangup detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)
The scenario is - system with TDM04B, a call comes in on a analogue line,
rings internally and then diverts to a
2020 Feb 20
0
anyone know of a list or wiki for GWC?
On Thursday, February 20, 2020 10:54:02 AM CST Fred Smith wrote:
> Hi!
>
> totally OT...
>
> Hoping there is a mailing list or wiki (or other help forum)
> for GWC, but haven't found one yet.
>
> I'm working on converting a bunch of my LPs to CDs, and am using
> GWC (Gnome Wave Cleaner, or GTK Wave Cleaner) to clean up the noise.
> It works great, but I
2003 Jun 28
0
SV: Newbie questions.....
Check to see if you can get a IOS code leverl that supports SIP on the
6500. then maybe you can use your E1 card directly. you can also get a
SIP version of the code for the 7960's etc
Dave
>>> jwi@weball.csis.dk 6/28/2003 2:56:12 PM >>>
Hi Chris
I've done a lot of things with Cisco AVVID solutions in the past.
> CallManager).....am I right in saying that Cisco
2009 Jan 07
2
OpenBSD TFTPD remap rules
Hi!
I am using OpenBSD 4.4 with the build-in TFTP-Daemon for PXE.
Distributing OpenBSD works fine.
Now I want to distribute OpenBSD and WinPE. I've installed
PXELinux 3.72 and configured it. Works also fine for PXELinux and
distributing OpenBSD.
To distribute WinPE I need a remap rule (\ -> /) for the TFTP-Daemon.
I've created a file /etc/tftpd.remap with the following rule:
rgG \\ /