Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers
Hi,
I know maybe this question is not related to asterisk, but I want to make
XML RPC web service to other http server.
I have elastix system. it is https and problem is source not destination
server. In xml rpc we have fsockopen connection to connect destination
server(xml rpc server). It return
On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer to
my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other side.
Receptionist
Hello,
thanks you for your answer.
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone ?
Jonas.
On 02/04/2013 02:29 PM, Steven Howes wrote:
On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
I call with my cellphone to our public telephone number
Our
On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone ?
Quick google doesn't turn up any results. Handsets probably dont support it.
Steve--
_
--
Hello,
and is there any setting in Asterisk to turn this functionality on/off ?
Maybe mine is not enabled.
Jonas
On 02/04/2013 03:30 PM, Steven Howes wrote:
On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone
One thing you can try is to set the following in your sip.conf.
sendrpid=pai
trustrpid=yes
You can put that on individual phone configurations in sip.conf or, as I
do, in a template that is applied to a set of phones.
I believe that was what I had to set so that the remote caller ID would
What is the PAI option below that you are talking about, for sendrpid ?
The manual only says that yes or no can be used..
On 2/4/13 9:39 AM, Kevin Larsen wrote:
One thing you can try is to set the following in your sip.conf.
sendrpid=pai
trustrpid=yes
You can put that on individual phone
According to the default sip.conf file:
sendrpid=yes ; If Remote-Party-ID should be sent (defaults to no)
sendrpid=rpid ; Use the Remote-Party-ID header to send the identity of
the remote party. This is identical to sendrpid=yes
sendrpid=pai ; Use the P-Asserted-Identity header to send the
On 3/2/13 4:59 pm, David Smiley wrote:
I finally found the perfect solution for me:http://www.amazon.com/La-**
Crosse-D111-101-E1-WGB-**Wireless-Monitor/dp/**
B0081UR76G/ref=dp_ob_title_defhttp://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def
The
You are correct, this is not an asterisk question. What I would suggest would
be to run your script outside of asterisk and debug the connection. Looking at
the php doc page for fsockopen
(http://php.net/manual/en/function.fsockopen.php), I see this example:
?php
$fp =
I don't get it, if it is a web service, why do you use sockets? Isn't it just a matter of calling the web service using curl,and then wait for the response? what am I missing?Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re:
Yes, I think curl would probably be a better option than trying to use sockets
directly, but if the socket won't connect it doesn't really matter what higher
level method is used.
-Justin
From: asterisk-users-boun...@lists.digium.com
On Monday 04 February 2013, Muhammad wrote:
Hi,
I know maybe this question is not related to asterisk, but I want to make
XML RPC web service to other http server.
I have elastix system. it is https and problem is source not destination
server. In xml rpc we have fsockopen connection to
I would just type in the web service url manually in a browser, and if the browser displays the response, then there it is, the connection to the host server is open.Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users]
Hi list,
I've always wanted to graph my active SIP calls on a map somehow, and
now I've finally taken the time to do it. My script is called VoIPGMap
and it displays active calls on Google Maps. Logic-wise it's designed
for a callthrough or calling card scenario, where calls will be
Check if you have selinux enforcing anf try to disable it
I am typing from my mobile phone...
Il giorno 04/feb/2013 18:43, C. Savinovich c.savinov...@itntelecom.com
ha scritto:
I would just type in the web service url manually in a browser, and if the
browser displays the response, then there
I have a recurrent problem on my asterisk box. I have VIA Samuel 2 as a
CPU. With asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled
from source.
I get a RED alrm drom the port 1( FXO) two or three times per day:
[Feb 4 15:54:57] WARNING[9991]: chan_dahdi.c:8018 handle_alarms:
On Mon, Feb 04, 2013 at 04:23:21PM -0500, neo haux wrote:
I have a recurrent problem on my asterisk box. I have VIA Samuel 2 as a
CPU. With asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled
from source.
I get a RED alrm drom the port 1( FXO) two or three times per day:
If your
If you're willing to spend a bit more you may also want to check out these
people -
http://avtech.com/
Mike
From: Carlos Alvarez [mailto:car...@televolve.com]
Sent: Monday, February 04, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How
We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our
streaming music on hold stopped working. I remember when we had first
installed 1.8 we had an issue where the streaming music on hold would not
work because Music On Hold was using the DAHDI timing module. We needed the
- Original Message -
From: Bob Pierce westman...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: g...@westmancom.com
Sent: Monday, February 4, 2013 6:14:26 PM
Subject: [asterisk-users] Asterisk 1.8 Streaming MOH timing
Client - Not for Profit in the Middle of the Jungle/Rain Forrest
Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding,
and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge
Podge of DYI wiring across remaining buildings. Phones - Total of about 50
extensions.
If you have the budget for two machines, run all services on one and keep
the other for a hot backup. Rsync the configs nightly. I'm guessing that
spare parts/repairs are far away from where you will be?
On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley jared.bax...@gmail.com wrote:
Client - Not
24 matches
Mail list logo
