Am 27.09.2014 17:28, schrieb d tbsky:
can someone give an example for the function? thanks for the help.
Not a programmer here, just grep -r'ed through the code, but maybe try
one of these:
G711A
G711_ALAW
--
_
--
Dear all,
My client has Asterisk based telephony system. He needs to add the intercom
feature in his telephones. He has 300 concurrent users with two PRI
Channels. I want to check if there is a possibility to have the requested
scenario by adding this feature to his current telephone system
Of course, it is possible. Depending on what the desired behavior is, it might suffice to enable
the auto-answer feature of an end point. You might also want to read about paging and intercom
for different scenarios.
jg
Dear all,
My client has Asterisk based telephony system. He needs to
2014-09-28 14:01 GMT+08:00 Markus unive...@truemetal.org:
Am 27.09.2014 17:28, schrieb d tbsky:
can someone give an example for the function? thanks for the help.
Not a programmer here, just grep -r'ed through the code, but maybe try one
of these:
G711A
G711_ALAW
thanks a lot for
How about recording the call calling it whatever you want, and then using a
custom AGI script to append the call to the original one? That’s how I would do
it if it were me.
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Hi All,
I have one asterisks server and 3 client (i'm using voip sip client for my
handphone).
I've configured sip.conf and extension.conf with 3 user different. And
nothing wrong with them, i could make them to make a call too.
what i want to ask is, i was try to use 1 user (ex:1001) in 2
On Sun, 28 Sep 2014, Anurag Rana wrote:
I am trying to record the call using MixMonitor.
...
Now I know that 'a' option is used to append the recording to a file but
I couldn't find any example on how to use it? Also if I use 'a' option
and file doesn't exist then is it created or it is
As the other posters said - try it!
Another option would be to use sox to combine files with some common
part of their filename.
On 28 September 2014 19:39, Steve Edwards asterisk@sedwards.com wrote:
On Sun, 28 Sep 2014, Anurag Rana wrote:
I am trying to record the call using MixMonitor.
check your ulimits :)
On 26 September 2014 17:15, CDR vene...@gmail.com wrote:
I am using Asterisk 12 svn, from today, and after a few thousand
calls, I run out of ports.
This happens eith PJSIOP and regular old SIP. I think it is RTP related.
Any idea how can I troblshoot this. It happened
hi,
anyone seen this?
actually this is my sip.conf file
[1002]
type = friend
context = test
username = 1002
secret = 12345
host = dynamic
if i want to make my client register to server with matching on username
instead of ip address, so the username is used just for 1 client how could
i do
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