Hello list,
Hope you all doing well!
We've been attempting to add a URI parameter to implement the trunk group
(tgrp and trunk-context) but discovered the add_uri_param() function only
works with constant string we can't use a pseudovar to inform the value
to be added. Anyone knows why such
Bounce.
Med vennlig hilsen
Pan B. Christensen
From: sr-users On Behalf Of Pan
Christensen
Sent: onsdag 13. juni 2018 09:19
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] outbound flow tokens and kamailio restart
Hello all.
I have created a WebRTC to SIP gateway. I implemented
Ok thats clear thanks for chekking, does the dlg_set_timeout support
$var?
Thanks,
Jan
Henning Westerholt schreef op 2018-06-14 16:17:
Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb
je...@cyberchaos.nl:
I do not see any errors in the logs, it looks like it just sets the
timer to 0 and
You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba wrote:
> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> > From the logs I see the jssip throw this error:
> >
> > "Failed to set remote offer sdp: Called with
Yeah, you need to set the correct offer, i did that a while ago, but i
can't remember how i did it.
Check out https://github.com/havfo/WEBRTC-to-SIP
Hope it help.
David
On Thu, Jun 14, 2018, 22:26 Daniel Tryba wrote:
> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> > From
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> From the logs I see the jssip throw this error:
>
> "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
>
> I would like to avoid RTPEngine, because from what I understand, FreeSwitch
> can handle the
Hi everybody, I followed this tutorial
https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamailio_33x_and_freeswitch_12x_for_media_services_and_sbc
And it works fantastic!
The next step was to add WebRTC support, so I added WebSockets module to
enable web clients to register
Daniel,
I appreciate the information. Than you very much.
Karthik
On Thu, Jun 14, 2018 at 1:18 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> you have to put this from the perspective of: changes to the SIP message
> (headers and body) are not immediately reflected. So even if you do a
>
Current Outbound Call:
WebRTC Client => Asterisk =>SIPDevice, here the SIPDevice communicates back
through Asterisk.
Desired Outbound Call:
WebRTC Client => Asterisk (which has an outbound_proxy set in pjsip)
=>Kamailio=>SIPDevice and back the same way?
The end goal being that the SIPDevice
Hi, I have a scenario where I am using.
t_save_lumps(), lookup() and then I can choose to drop some branches in
branch routes.
However, it is possible that I endup without any branch left, in this case
t_relay() is returning and no failure route is called/created.
At this point I need to recover
On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote:
> If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio
> actually make the SIP call?
Ehhh, yes. Why wouldn't that be possible?
> At some point I would like outbound calls to be controlled by Kamailio
> so
Hi Henning,
Thank you a lot!
With kind regards,
Jurijs
On Thu, Jun 14, 2018 at 5:02 PM, Henning Westerholt wrote:
> Am Donnerstag, 14. Juni 2018, 08:31:58 CEST schrieb Jurijs Ivolga:
> > Thank you a lot for your input.
> >
> > But I was asking if there is a point to create patch from this 2
Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb je...@cyberchaos.nl:
> I do not see any errors in the logs, it looks like it just sets the
> timer to 0 and disconnects the call.
Hello Jan,
I just looked quickly in the code, it seems that the timeout value is not
interpreted as
Am Donnerstag, 14. Juni 2018, 08:31:58 CEST schrieb Jurijs Ivolga:
> Thank you a lot for your input.
>
> But I was asking if there is a point to create patch from this 2 commits
> and apply to 4.4. Is it worth? Or there is no way to make this work
> properly on 4.4? As I see, some part of code
Good Morning All!
If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio
actually make the SIP call?
At some point I would like outbound calls to be controlled by Kamailio so that
the outside endpoints never communicate with the PBX.
Currently a call goes through Kamailio
Hi,
could you share some details, how you solved the problem?
Best Regards
Markus
On Thu, Jun 14, 2018 at 11:30 AM, eyas barhouk wrote:
> solved dears
> thanks
>
>
>
> Sent from my Samsung Galaxy smartphone.
>
>
> Original message
> From: eyas barhouk
> Date: 14/06/2018
solved dears
thanks
Sent from my Samsung Galaxy smartphone.
Original message
From: eyas barhouk
Date: 14/06/2018 2:23 am (GMT+02:00)
To: sr-users@lists.kamailio.org
Subject: [SR-Users] integrate huawei HSS with kamailio
hello dears
i'm trying to integrate kamailio IMS
solved in the same way dear Carsten
thank you for your kind help
Sent from my Samsung Galaxy smartphone.
Original message
From: Carsten Bock
Date: 14/06/2018 12:23 pm (GMT+02:00)
To: eyas barhouk
Cc: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] Forbidden
Hi,
you need to look into the data provisioned in the HSS.
You should have an IMPI as well as an associated IMPU like this:
"09876993998754@IMS1.NET1"
We've tested Kamailio with a whole bunch of HSS's (from NSN, E///,
ZTE, SummaNetworks, ...), so it's definitely not an issue on Kamailio.
Huawei
Hi Henning,
Thank you a lot for your input.
But I was asking if there is a point to create patch from this 2 commits
and apply to 4.4. Is it worth? Or there is no way to make this work
properly on 4.4? As I see, some part of code what is touched by this 2
commits differs quite a lot, so I'm bit
Hello,
you have to put this from the perspective of: changes to the SIP message
(headers and body) are not immediately reflected. So even if you do a
replace or subst operation, changes are not visible. If you do
remove_hf() or append_hf(), it happens the same.
The FAQ has an entry for it:
-
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