Displaying 20 results from an estimated 90 matches similar to: "No subject"
2013 Jun 28
0
No subject
<br>
[memberconnector]<br>
;<br>
exten =3D> _XXX,1,Dial(SIP/${peerPrefix}$<u></u>{EXTEN},${TIMERINGQUEUE}=
,)<br>
=A0 =A0 =A0 same =3D> n,NoOp(DIALSTATUS=3D${<u></u>DIALSTATUS})<br>
<br>
As you can see, all status are empty,<div class=3D""><div><br>
<br>
-- <br>
2011 Apr 12
0
No subject
r>
<h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010=
)
</h2>With SIP 3.2.X firmware (available on the Polycom download site)=20
and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20
showing statuses of Ringing, Inuse and Online and one touch directed=20
call pickup.
<br>On the asterisk side all that needs to be done is to add a hint
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2011 Sep 02
0
No subject
crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the "gdb asterisk
<corefilename>", and
get a stack trace. That should give us some idea of what happened.
>
> I have a fairly simple Followme sequence in place to see how it works
> before I get into the complex scenarios.
2018 Aug 29
1
Exit mailings
And me too, please
[Smile]
From: Hub. Haenen<mailto:haenenhub at hotmail.com>
Sent: Wednesday, August 29, 2018 5:11 PM
To: Icecast at xiph.org<mailto:Icecast at xiph.org>
Subject: [Icecast] Exit mailings
Could you please take me of the mailinglist of Icecast. Thanks!
Greetz,
Hub.
<http://nl.linkedin.com/in/hubhaenen>
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An HTML
2009 Dec 14
1
Rewrite calling number of incoming call
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite "shows" 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten => 977/733025975,1,Set(CALLERID(number)=0317998975)
exten => 977,n,Dial(SIP/0317998977)
[Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272
2011 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web =
interface.
Let=92s say Yves=92 =93special conference=94 is 5555. The moderator =
would start
using this command
Exten =3D> s,1,meetme(5555)
The participants would do
Exten =3D>
2015 Oct 23
0
icecast
hi all
how do i editing line that has to be uncommented so that the server starts??
<icecast>
<!-- location and admin are two arbitrary strings that are e.g. visible
on the server info page of the icecast web interface
(server_version.xsl). -->
<location>Earth</location>
<admin>icemaster at localhost</admin>
<limits>
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2009 Jul 20
0
No subject
mailboxes).
Are you certain that removing either 612 or 610 mailbox would keep Asterisk
from complaining ?
>
> However, the MWI does not indicate voice mails for 610 and I keep seeing
> this error message:
>
> ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
> 610 in context a10
>
> However, mailbox 610 is clearly defined in voicemail.conf:
>
2015 Oct 23
2
icecast
Lol... It has started! But it appears that the username had to be 'source' as Phillppe said!
Sent from my iPhone
> On Oct 23, 2015, at 14:19, Alan Bowness <awi3 at live.com> wrote:
>
> maybe wrong english words are being used?
>
> I have a feeling that the default settings are at fault here, in the .xml file, there is a line that has to be uncommented so that the
2009 Jul 20
0
No subject
-- SIP/ vaso -e26c answered Zap/14-1
-- Executing DumpChan("SIP/ vaso -e26c", "") in new stack
-- Executing DumpChan("SIP/vaso-e26c", "") in new stack
Dumping Info For Channel: SIP/vaso-e26c:
============================================================================
====
Info:
Name= SIP/vaso-e26c
Type=
2016 Jul 22
0
Config and other areas of struggle
>Hi Alan,
Best direct it to: Damien
Hi... anyway
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2016 Jul 23
0
Config and other areas of struggle
Hi Alan,
Yes, this is what I am saying. The ports are already forwarded. Still no go.
Cheers.
From: Alan Bowness
Sent: Saturday, July 23, 2016 10:01 AM
To: Icecast streaming server user discussions
Subject: Re: [Icecast] Config and other areas of struggle
Confused me for ages.. until one day!
Which out of the listed machines (on the router) is the one you are needing the port forwarded to?
2011 Sep 02
0
No subject
typing his number, though there is a 15 seconds timeout, and even if I type
the number very fast it still may happen to me.
*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
In the logs:
When it fails:
- - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr')
- - User disconnected
When it succeeds:
- - <SIP/ipbx-iwred-000002e> Playing
2009 Jul 20
0
No subject
in which Dial originally occurred, but for an unknown reason, it can't find
the appropriate hook to keep on.
Do you have any working sample ?
Regards
--0016e646050485a6cf0474456758
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable
Hello,<br><br>I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find=
a way to successfully
2016 Jul 23
1
Config and other areas of struggle
What are you using to forward the 'music' to the Icecast server that I can see? If you are using darkice for example, have you configured the darkice.cfg so that it matches icecast's configuration? Icecast appears to be working, but has no stream.
If I knew what the passwords were, I could configure my darkice on my pi to forward music to your Icecast and it would stream it - you just
2004 May 04
2
Can Asterisk support R2 signaling
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
>From: asterisk-users-request@lists.digium.com
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
>Send
2016 Jul 23
2
Config and other areas of struggle
Confused me for ages.. until one day!
Which out of the listed machines (on the router) is the one you are needing the port forwarded to?
Forward that machine to the port you require, I would have used a lower port number 8010 or another unused port.
From: Damien Sykes-Lindley
Sent: Saturday, July 23, 2016 9:39 AM
To: Icecast streaming server user discussions
Subject: Re: [Icecast] Config
2005 Jul 10
7
QOS HELP PLEASE
ive got problems with my network (120 people)
ive got big pings (300ms)m whereas there are normally about 19ms.
i do not know if my qos is proper (fast i mean).
www.tdi.pozman.pl/fir2 - my qos
www.tdi.pozman.pl/rules - my firewall
can sb tell me if do it ok ?
--
*Dariusz ''tdi'' Dwornikowski | Gentoo | admin at pozman.pl |