Displaying 20 results from an estimated 10000 matches similar to: "Separating Asterisk SIP extensions from dialing each other."
2006 Apr 01
2
TO have ringing tone instead MOH
I need to avoid MOH on my asterisk box, so i need to have a ringing tone
when attendant transfer is made, or a call is on hold..
Is there any way to do that.
I did not see a simple way to do that.
Regards
2006 Nov 17
5
spc.exe
Does anyone have a copy of spc.exe they could send me?
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2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2006 Jun 14
1
SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the web
interface regarding this feature. I just want callers to hear the busy
tone when the called party is at the phone. Probably I can accomplish
this by using the "disable call waiting" in asterisk as well, but I have
not been able to find any
2006 Apr 21
1
1.2.7.1 on FC5 won't make install
The make seems to go okay.
[root@somebox asterisk-1.2.7.1]# uname -a
Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux
mkdir -p /var/lib/asterisk/sounds/digits
mkdir -p /var/lib/asterisk/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if grep -q "^%`basename $x`%" sounds.txt; then \
install -m 644 $x
2006 Jun 09
3
Compiling SVN Trunk
I have the same problem on some modules.
For example app_math.so
[app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728
__load_resource: missing mod_data for app_math.so
Any help?. I have been looking , but nothing reasonable found.
Thanks
--
Alberto Sagredo
2006 Mar 23
3
Polycom 501's for sale
Converted a strictly VOIP system in NYC to NEC IPK TDM system...
will have 25 Polycom 501's for sale.
Best offer, offlist only please.
R
2006 Apr 10
5
SPA-941/942 Bulk provisioning
Has anyone got any information on bulk provisioning of Linksys SPA-941/94s?
There is an overview in the admin guide but it refers to a different
provisioning guide that I haven't found anywhere.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>
kerryg@techdatapros.com
2006 Jan 12
2
Easy to Access Telephone Directory AGI
I've written myself a easy to use telephone directory
which I use at home and thought it may be of interrest
to others.
The purpose of this agi script is to provide an online
telephone directory that can be easily accessed using
the numbers on the phone dial pad.
You select entries by spelling out the name of the
person you want to contact using the phone dial pad.
Now this is normally
2018 Nov 15
7
Queue not dialing out to cell phone for some reason
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and
2006 Mar 29
3
SMS in Spain (it seems Protocol 2)
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
I am trying to send SMS in Spain using landlines. It seems that
app_sms.c only handles Protocol 1, but Spain and Italy are using
Protocol 2.
I have been searching in Internet without any results... anybody is
sending SMS from Asterisk (or any method) using Protocol 2? (so, it
seems,
2003 Nov 01
13
Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I prefer not to go
back to RH7.3...
BTW, where would I find a useful FM?
David
--
David J. Sussman, MBA
email:
2012 Feb 16
2
Asterisk && RTCP
Hello list,
I need to know about Asterisk's friendly nature with RTCP. I've phones
which support RTCP and they connect to the outer world via multiple
carriers. In one of my recent packet traces I've observed that the caller
initiated a call with rtcp string in SDP while for the same
call dialling our from Asterisk to the carrier has no RTCP string in SDP !
Can anyone please tell why
2006 Jun 28
1
Work required - modify Asterisk + SEMS
Hi all,
I am looking for a developer or developers that can implement the following:
- Modify an Asterisk server in order to support one inbound RTP and
several outbound RTPs, I was thinking SEMS may provide a very good
starting point. The idea is to make a PA system over IP. We do *not*
want full-duplex audio.
- Implement a client in Qt/C++, that allows to send audio to this
platform, and
2008 May 12
2
Which sound file formats?
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so will only
have SIP connections.
The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729,
.gsm, .ulaw, and .wav.
What are the minimal
2005 Mar 08
0
Could dialing long extensions be a problem?
I have two DIDs from NuFone.
I have for the first DID a short extension (601 = 3 digits) and on the
other one a long extension (886212345678 =12 digits)
Extensions.conf looks like:
[fromNuFone]
exten => 888xxxxxxx,1,Dial(SIP/601,20,tr)
exten => 888xxxxxxx,2,Dial(SIP/602,20,tr)
exten => 888xxxxxxx,3,hangup
;
;
exten => 866xxxxxxx,1,Dial(SIP/886212345678,60,tr)
exten =>
2012 Feb 22
2
codec mismatch on channel
Hi
I am keep getting this warning message when doing attendant transfer:
WARNING[6027] channel.c: Codec mismatch on channel
Local/XX at Inside-1f32;1 setting write format to slin from alaw native
formats (alaw)
What can I do to lose it.
I am using asterisk 10.1.2
Best regards
2009 Sep 29
2
dialing 0 in directory()
I've got a context in my dialplan like so but pressing 0 doesn't seem
to be working. Instead of dropping out to the "o" extension, it's
just returning to the start of the direcotry app. Same with star.
Anyone see where I've gone awry?
[attendant]
; <snip>
exten => *,1,NoOp(Attendant: Directory)
exten => *,n,Directory(default,attendant,eb)
2002 May 23
3
separating a digest into separate messages
Could someone suggest a mail reader that will parse a digest from r-help
into separate messages?
Preferably, this reader would allow replies or forwards of separate
messages. I use pine on a Sun running SunOS 5.7. If there were an add-on to
pine that would accomplish this task, that would be even better.
Anne
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Anne E. York
National Marine Mammal Laboratory
2005 Jun 07
3
Icecast RTP support
Hi,
Does Icecast has RTP support for streaming OGG/Vorbis and OGG/theora
media files?
If so, then please give me some pointers on how to configure my
Icecast server to listen to RTP requests.
If it's not available, then is the inclusion of this feature, in plans
for future release of the product?
Thanks
--
Subhabrata Bhattacharya