Hi,
Sorry to insist, but I still not have any solution. Does anybody have an
idea ?
Thanks!
2011/4/20 John Alexis kasteris...@gmail.com
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load
iax accounts.
Settings seems fine
Is there a way do what is sometimes called a 3rd party transfer in
Asterisk. That is; Call A comes in and is answered B. B then places A
on hold and calls C. After C answers, BC chat for a moment, then B
brings A on line. After making intro's B then drops off call.
Thanks,
Jeff
On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson
jjohn...@neturallyspeaking.com wrote:
Is there a way do what is sometimes called a 3rd party transfer in
Asterisk. That is; Call A comes in and is answered B. B then places A on
hold and calls C. After C answers, BC chat for a moment, then B
Op 22-04-11 22:58, Steve Edwards schreef:
Op 22-04-11 18:13, Eric Wieling schreef:
sip set debug on should help
On Fri, 22 Apr 2011, Paul van der Vlis wrote:
I've tried it, but no, nothing...
Sounds like you have very basic network issues.
Can this host ping your SIP endpoint?
I'll give it a whirl,
Thanks,
Jeff
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Saturday, April 23, 2011 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Op 22-04-11 23:49, Jamie A. Stapleton schreef:
I can see your server just fine...
-bash-3.2# ./svmap.py xen8.vandervlis.nl
| SIP Device | User Agent | Fingerprint |
--
| 91.198.178.28:5060 |
Hello,
I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple
problems with DTMF.
I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does
IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two
servers communicate via SIP with
Hi.
Im having trouble setting variables in channel dialplan and re-using them in
Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context:
Hi,
Using DumpChan(); Seems that Channel (where the call goes first) is a
sub-channel of Context/Extension (where the call goes on CONNECT) ??
first I have:
Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2:
Then after:
Dumping Info For Channel:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID:
Where would one find, or better yet determine from code, all of the table
definitions for ARA dynamic families?
There seems to be some bits and pieces in various places around the internet,
ie. voip-info, the definitive guide, ect. but nothing complete or definitive.
I have wondered about this
On Sat, 2011-04-23 at 10:52 -0700, Jason Rogers wrote:
Where would one find, or better yet determine from code, all of the table
definitions for ARA dynamic families?
There seems to be some bits and pieces in various places around the internet,
ie. voip-info, the definitive guide, ect. but
Hello asterisk asterisk,
Am 2011-04-23 06:24:36, hacktest Du folgendes herunter:
Look at this wiki for help.
http://wiki.e1550.mobi/doku.php
For asterisk, you can use your USB stick for voice/SMS but not internet at
the same time. A separate internet connect is required per my
If you don't see the call coming in when you have sip debug enabled, then the
call is not making it to your server.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Friday, April 22,
check this
http://www.voip-info.org/wiki/view/Asterisk+sip+nat
On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear * users,
in your opinion, when using a * as a public server, is good practice
enabling nat=yes in sip.conf for all the peers?
Can
check this url, let me know if any problem
http://www.voip-info.org/wiki/view/Asterisk+video
http://www.voip-info.org/wiki/view/Asterisk+video
http://www.voip-info.org/wiki/view/Asterisk+videobest
On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies davies...@gmail.com wrote:
Hi,
Can anyone let
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