Displaying 20 results from an estimated 8000 matches similar to: "How to read or relay SIP PUBLISH messages ?"
2017 Feb 16
2
How to read or relay SIP PUBLISH messages ?
2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>:
> On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote:
> > Hello,
> >
> > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP
> > hardphone.
> >
> > When a phone has enabled this feature, it would send a SIP PUBLISH to its
> > SIP Server letting this server dispatch to
2015 Mar 12
1
PJSIP and Kamailio without registration
From: Matthew Jordan <mjordan at digium.com>
>
>
> >> If the INVITE request is not shown in the CLI with 'pjsip set logger
> >> on', then Asterisk is not actually receiving the request.
> >>
> >> Does a pcap show the message being sent to the correct IP/port? If you
> >> change the transports to bind to port 5060, does that change
2015 Apr 20
1
Kamallio registration
Hi Guys
Is it possible to register Kamallio directly to our SIP provider then load
balance the RTP to 2 asterisk servers?
We cant do the registration from the asterisk boxes as we want to do it
directly from Kamallio.
Is this possible?
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2009 Mar 06
1
Asterisk and sip router integration
Hi,
Does anyone have some good examples of a Kamalio or OpenSips
configuration that integrates with Asterisk?
Essentially I want to use the SIP router as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've looked for examples on the project web sites, but I haven't found
anything decent yet.
Thanks.
-- James
2011 Mar 24
1
Linux Based Billing and CDR
Hi All,
Do you'll have any recommendations on a Linux based Customer Management and
Pre-paid Billing system for Asterisk, Freeswitch or Kamalio?
The system should also allow customers to register, login, buy more credit,
view call records, etc.
Commercial or Open-source are ok as long as they run on Linux.
Thanks,
A.
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2009 Jul 20
0
No subject
used Kamalio to "supplement" the features that Asterisk either doesn't
provide or doesn't provide in as nice a form as the OP desired - can't
really speak beyond this as I am not one of them.
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2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2009 Mar 24
1
Relay Register
Good morning everybody.
My question is simple.
Is there a way to perform relay register with Asterisk ?
More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk :
REGISTER REGISTER
Client ------------> Asterisk ---------------> OpenSIPS
So Asterisk keep a list of registered clients and only allows them to
2020 Jan 15
4
Asterisk16 - PJSIP - Error 401 on outbound registration
Hi all,
we face a strange behavior while connecting an Asterisk16 instance with
PJSIP to 2 providers: we receive error 401 Unauthorized, both of them
having Kamailio as front-end. With other providers -we don't know if
they run kamailio- registration is just fine.
One of the provider took a pcap and told us that expiration was set to 0
that's why they don't accept the
2003 Nov 18
1
Will Asterisk be supporting RTCP XR in the future?
This article below came up on the newwire. The RTCP XR RFC was published.
Will Asterisk be supporting this function in a future release? Does anyone
know if any phone vendors are going to be supporting it?
Thanks
Lee Goodman
Our Technology Update this week is about one of those
mechanisms. Known as RTP Control Protocol Reporting Extensions
(RTCP XR), the technology defines a standard way to
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video
2015 Mar 04
2
WebRTC phone
For those that were interested I have attached the kamailio.cfg which we
have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
following yum packages:
kamailio.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-auth-ephemeral.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-bdb.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
2015 Feb 26
2
WebRTC phone
Can anyone recommend a good WebRTC phone to use with Asterisk? I do
not mind if it is commercial or open source. Customers are starting to
ask for web solutions and we need to start testing.
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez
+52 (55)9116-91161
2015 Jan 29
2
any valid up-to-date info about Kamailio-Asterisk integration ?
Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N
instances of Asterisk
Are there any
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (
2015 Jan 21
1
PJ SIP realtime with Kamailio / opensips
Hi all,
I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.
All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.
In order to do what Matt suggested would I be correct in assuming I would
have to use the
2010 May 17
1
R: new way of asterisk and kamailio(openser) realtime integration
Works for me....
Thanks,
Hristo Benev
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc
Sent: Monday, May 17, 2010 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration
2014 Feb 20
2
How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration
(MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be
2015 Feb 13
2
Debugging some DTMF Weirdness.
I'm attempting to find where my extra long DTMF Tones are coming from.
I'm dialing from my sip handset through my proxy to my Asterisk box which
is my PSTN Gateway.
I'm pressing 4 to select a menu and everything is fine.
[Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on
SIP/trunk-0a02dee0
[Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough
2014 Jan 20
3
Asterisk not receiving call from VPN address
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and forwards
it to the Asterisk server on it's 103.x address, Asterisk never sees the
call.
If Kamailio receives a call from the VPN and forwards the call to the
Asterisk server on it's 172.x