On Thu, Oct 25, 2018 at 4:32 AM Olivier wrote:
> Hello,
>
> I'm testing an Asterisk instance.
> At the moment, I'm focusing on its capability to receive and challenge
> incoming SIP Registrations.
>
>
If all you want to do is test inbound registrations you can find an example
SIPp scenario in the
Hello,
I'm setting up a new cluster that must replace several old Asterisk
instances.
For various reasons, this new cluster must use chan_sip (migration to PJSIP
is planned in a later phase).
This new cluster uses VRRP in active/passive mode:
- at any time, only one cluster member is active,
- wh
On Thu, Oct 25, 2018 at 6:58 AM marek cervenka wrote:
> hi,
>
> i have webrtc client chrome69/jssip which is connecting to asterisk
> 13.23.1/pjsip
>
> i have strange problem where pjsip aor stays in status "created"
>
> sip trace on asterisk looks ok.
>
>
> do you think if this can be bug?
>
It
hi,
i have webrtc client chrome69/jssip which is connecting to asterisk
13.23.1/pjsip
i have strange problem where pjsip aor stays in status "created"
sip trace on asterisk looks ok.
do you think if this can be bug?
test*CLI> pjsip show aors
Aor:
Contact:
=
Would really appreciate some help here - into day 4 of trying to bridge a
PJSIP call to an existing confbridge.
There's a fair amount of dialplan and log to show which doesn't really work
well via plain text email, so I've taken it over to the forum at
https://community.asterisk.org/t/bridging-an-
Hello,
I'm testing an Asterisk instance.
At the moment, I'm focusing on its capability to receive and challenge
incoming SIP Registrations.
For various reasons, I would prefer to use SIPp instead of Asterisk to act
as SIP Client.
Has someone successfully done this ?
If negative, what explains th
