rt to the stable releases too.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 01:45, Julian Kay wrote:
> thanks, I appreciate that!!!
>
> Juls
>
> -Original Message-
> From: users-boun...@lists.opensips.or
-solutions.com
On 14.01.2016 00:28, Julian Kay wrote:
> HI,
>
> I really appreciate you taking the time!!
>
> yes OpenSIPS is connecting to Tomcat, the problem I see (I think) is when the
> event is being raised OpenSIPS is NOT sending the complete URI. Tomcat server
> re
ttp://www.opensips-solutions.com
On 13.01.2016 03:02, Julian Kay wrote:
> Hi;
>
> Thanks for the help!
>
> the event subscription seems to be successful as show from the output
> of a test xmlrpc test client
> Event:: E_SIP_MESSAGE id=11
Subscriber::
> socket=xmlrpc:192.168
sips.org/Documentation/Interface-CoreMI-1-11#toc18
After that, when events happens, check at network level if there is any attempt
from opensips side to deliver the event via XMLRPC to the indicated URL.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-soluti
s.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Julian Kay
Sent: Thursday, January 07, 2016 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP events
Thanks for your help!!
yes the event is reaching the xmlrpc server
sure you are raising that event from your script (i.e.
raise_event("E_SIP_MESSAGE")).
http://www.opensips.org/Documentation/Tutorials-EventInterface#toc9
Best regards,
Răzvan
On 12/24/2015 03:39 AM, Julian Kay wrote:
Thanks for the input, but for now I need to work with XMLRPC.
is
n
subscribe as reader of the queue and do whatever task needs to be done. You can
go as far as using this CDR as well.
Goodluck,
Tito
On Wed, Dec 16, 2015 at 6:14 PM, Julian Kay wrote:
Hi;
I'm looking for some guidance or confirmation that I'm on the right path. If I
Hi;
I'm looking for some guidance or confirmation that I'm on the right path. If
I want an external app to be able to listen to SIP events like SIP event 180
(ringing), is registering a custom event with OpenSIPS MI and then use
"raise_event" to fire a custom event a good way to get my external
On 17.08.2015 08:14, Julian Kay wrote:
Hi;
I'm looking for information on monitoring extension status on phones registered
with OpenSIPS.
I'm looking into interfacing a java web applet to OpenSIPS.
If anyone knows where I may find helpful info I wou
Hi;
I'm looking for information on monitoring extension status on phones
registered with OpenSIPS.
I'm looking into interfacing a java web applet to OpenSIPS.
If anyone knows where I may find helpful info I would appreciate it.
Thx!
JK
s... but the sending is stumping me.
Thx
Juls
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Julian Kay
Sent: Sunday, June 21, 2015 1:24 AM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] outbound routing
Hi;
Hi;
I've been looking for a while, but I haven't been able to find too much info
on routing outbound called through a SIP Trunk... can someone point me to
where I can find info / examples / tutorials / etc.
I would really appreciate some direction, and sorry if it's an elementary
grade que
Hi
Can anyone give some ideas why connections made through Opensips after 20 /
30 seconds loses audio and soon after (seconds) disconnects?
Thx!!
Juls
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Users@lists.opensips.org
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peer name in sip.conf
As per the sip trace
Opensips gets an INVITE
FROM "Anonymous"
TO
Thanks, I do appreciate the input very much
Juls
From: Stefano Pisani [mailto:stefano.m.pis...@gmail.com] On Behalf Of
Stefano Pisani
Sent: Sunday, May 10, 2015 2:37 PM
To:
ject: Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs
Just define a peer in sip.conf
then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple.
regards,
s
Il 07/05/2015 08:32, Julian Kay ha scritto:
Hello Everyone;
I want to be able to forward calls from Asterisk t
Hello Everyone;
I want to be able to forward calls from Asterisk to a phone registered with
OpenSIPs, can anyone give me some information on how this can be done? Or a
link to the info I would need to do this.
Thanks for any help
Juls
___
Users
I've been trying to send calls from OpenSIPS to Asterisk using the
ds_select_dst command. I'm able to do it with a 3CX soft-phone but all my
Snom phones fail, they are all on the same network (Snom 300, 320, 360).
Can anyone suggest what I'm doing wrong or what I can try?
Asterisk returns SIP
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