Hello guys,
there is something i'm not sure and would like to ask please :
let's say i have B calling and talking to A.( A -- B )
Then B would like to attended transfer A to C.
In the behind extensions.conf when B is calling C (first part of the
attended transfer) the Dial
Hi Sammy.
Thank you very much for you help!
Answering your questions:
a. I don't know almost anything about SIP Proxies. I just knew about Asterisk.
But, I will investigate about SIP Proxies today, thankful your citation.
b. We have already built a system that uses Asterisk, RTSP and provide
You could initiate a local channel – one leg enters to the conf. room and
another to application playback to play the file you wanted.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, July 14, 2015 1:08 PM
To:
Dan Cropp wrote:
Thank you Joshua.
That did in fact solve the problem I was seeing.
I am now experiencing another issue. The 3rd party sends their messages from
various port numbers.
However they only read messages sent to them on port 5060.
For example, right now I receive the first INVITE
I would like to figure out using confbridge how
to play a file after the conf is built.
not really a per user thing - just conf is up and ready
and need to play a file to all in the conference.
(I am creating my conf on the fly and bringing in other
devices on the fly).
How can that be
Thank you Joshua.
That did in fact solve the problem I was seeing.
I am now experiencing another issue. The 3rd party sends their messages from
various port numbers.
However they only read messages sent to them on port 5060.
For example, right now I receive the first INVITE with port 1234.
I am currently running Asterisk 13.1.0-1
I have a chan_sip configuration that works fine with a 3rd party. Third party
does not use authentication or registration, it's ip based authentication...
When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk
side.
What has me
Dan Cropp wrote:
snip
My pjsip.conf looks like…
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[3400]
type = aor
max_contacts = 1
remove_existing = yes
contact=sip:xxx.xxx.xxx.xxx
[3400]
type = endpoint
context = DEF
transport = transport1
aors = 3400
accountcode =
