From: virus.c...@mail.ru virus.c...@mail.ru
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
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To: asterisk-users@lists.digium.com
Subject: [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40
Date: Tue, 07 May 2013 07:53:53 +0600
help
2013/5/7 Matthew Jordan mjor...@digium.com
On 05/06/2013 05:54 PM, Olivier wrote:
Hi,
Before trying to script res-memcached installation (see res_memcached
https://github.com/drivefast/asterisk-res_memcached), I banged into
this on a fresh 11.3.0 setup:
snip
My questions are:
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone
Answering my own question. Setting the following in alsa.conf fixed my
problem:
input_device=plughw:0,0
output_device=null
Changing the input device to plughw helped some but didn't completely clear
the audio up. Setting the output device to null did the trick. I'm
wondering if there was some
You can use POE for geting AMI events.
I'm sending you a simple poe.pl file in attachment, where you will get all raw
events, and some callbacks are implemented for particular events.
For your case you can add few callback like conference join event, conference
leave event.
Muhammad Faheem
On 05/07/2013 05:13 AM, Olivier wrote:
2013/5/7 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com
2. It appears as if you're running a modified version of Asterisk, in
which case all bets are off. This works fine on the Linux build
agents,
which is what we use to
2013/5/7 Jason Parker jpar...@digium.com
On 05/07/2013 05:13 AM, Olivier wrote:
2013/5/7 Matthew Jordan mjor...@digium.com
2. It appears as if you're running a modified version of Asterisk, in
which case all bets are off. This works fine on the Linux build agents,
which is what we use
Hi,
I hope you might give me some hints on how to find where my
configuration is wrong, I am new to Asterisk and do not know, how to
find the problem.
Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the
same maschine: Hylafax fax server. I want hylafax to use t38modem (a
Am 07.05.2013 18:23, schrieb Sebastian Niehaus:
For some reason, t38modem tells hylafax the line is BUSY so there is no
fax send.
Well, I may add the log of t38modem (sorry for the ugly formatting)
Parts I consider as most important are:
ModemConnection::SetUpConnection dstNum=189659
Hello;
What is the best method to let the voice quality through Dahdi channels to be
clear and no echo? Is it the wanpipe or it is working only with sangoma?
Regards
Bilal
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SN == Sebastian Niehaus nieh...@web.de writes:
SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the
SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a
SN virtual T.38 modem) for sending faxes. t38modem schould connect to
SN asterisk on the same host.
SN
On 5/7/13, Johann Steinwendtner steinwendt...@gmx.net wrote:
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to
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