Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib
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Hi Friend
I first store voicemail with video using Xlite.
Xlite has problem that for storing video ,required to click on start
buttom.And when i retreive it, then i find that , video and voice mail have no
syncronization.As Example--I record voice 5 sec and after 5 second then click
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
On 01/02/2012 11:21 AM, sean darcy wrote:
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
Hello,
Can anyone explain what this attack was trying to do? *19.19.19.19 *is my
server IP and it seems that they are trying to use my server IP to initiate
a SIP call to 199.16.208.29 or 199.16.208.30. Is that so?
*Call Date Channel Source CLID
Please help, I have tried many things I cannt make it work, when I make an
audio call it is converted by asterisk to video call request, Please how to set
the call type at extensions.conf, I tried setting the codec manually but didnt
work also... any help .. any suggest will be great
Thanx
We get the same error with this version.
On Sun, Jan 1, 2012 at 6:13 PM, Matt Hamilton mistral9...@hotmail.comwrote:
I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 -
both on the same subnet, no nat.
The following is the flow of messages:
1. UAC sends the registration
On 01/02/2012 11:30 AM, sean darcy wrote:
On 01/02/2012 11:21 AM, sean darcy wrote:
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
Faraj Khasib wrote:
Please help, I have tried many things I cannt make it work, when I make an
audio call it is converted by asterisk to video call request
Not that I can help, since I don't do any video calling.
But, if you don't give any information about your system (OS and
version,
Hi all,
I've got a batch of Polycom 335's that I'm trying to get setup. The phone
works fine, but when I plug a PC into the PC port on the back, the PC can't get
to the Internet.
I've turned off all of the VLAN configuration. I've never had this problem
before, so I'm at a loss as to how
Mike Diehl wrote:
Usually, it just works...
Any ideas?
I've seen this before.
One of our facilities have 'smart or managed' switches that have caused
no ends of problems, including preventing computers plugged into the
phones not having network access.
You may want to review your
Agreed. Check the switch for some kind of port security. Most of the time this
would disable the interface if more than one MAC is present but you never know.
Are there blinky lights on the pc?
Also if provisioning via some sort of server check the MAC-boot log that the
pgone uploads.
Good
On Wed, Dec 28, 2011 at 3:32 PM, Gilberto Verástegui gilbert...@ti-m.com.mx
wrote:
Calls to long distance get disconnected before answer.
Telco: Alestra
Country: Mexico
System: Elastix 2.2
Digital Card: Digium TE122
Log:
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: --
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my
clients but the request is recieved at the other client as video call request
since I am enabling video support for sip
Sent from my iPhone
On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, Doug Lytle supp...@drdos.info wrote:
Faraj
Hi,
Please give you sip phone name and sip.conf and extensions.conf details
which is using for that communication.
And CLI output of asterisk is also required.
On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib fkha...@iconnecths.com wrote:
I use asterisk 1.6, my clients are sip clients, I dail
Which is?! What I am missing how to set dail plan in extension.conf to pass
call type as its Not convert request to video
Sent from my iPhone
On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, virendra bhati
virbh...@gmail.commailto:virbh...@gmail.com wrote:
Hi,
Please give you sip phone name and sip.conf and
Which is means like if you are using sip 1234 then give the details of
[1234] into that open thread and relevent extensions details too
On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib fkha...@iconnecths.comwrote:
Which is?! What I am missing how to set dail plan in extension.conf to
pass call
Here is the thing, my sip client can call the same. Extension once as audio and
once as video, so I cannt turn off video supportat reciever, what I guess can
be done is in extension.conf , there must be flag or something I can manipulate
...
Sent from my iPhone
On ٠٣/٠١/٢٠١٢, at ٨:١٩ ص,
Hi
Might be it will help. Read it and set in extension as per your need.
core show function CHANNEL
-= Info about function 'CHANNEL' =-
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/sets various pieces of information about the channel, additional
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