Displaying 20 results from an estimated 171 matches for "kamailio".
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...
2015 Mar 04
0
WebRTC phone
On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote:
> 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
>...
2015 Mar 09
1
PJSIP and Kamailio without registration
Hi,
I want to have Kamailio in front of one or more Asterisk boxes.
I don't think it is necessary for Kamailio and Asterisk to register with
one another. I'd like for PJSIP to recognise Kamailio by its IP address.
I have two boxes, both have public IP addresses, they also have private IP
addresses and can communicat...
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 Ast...
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
2010 May 17
1
R: new way of asterisk and kamailio(openser) realtime integration
...iginal 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
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Daniel-Constantin Mierla
Inviato: luned? 1...
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 aster...
2010 May 17
1
new way of asterisk and kamailio (openser) realtime integration
Hello,
I put together a new tutorial about asterisk realtime integration with
kamailio (openser). This time the database used is the one of asterisk,
also call routing logic is controlled by asterisk, here is the link:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
Practically is an easier way to scale starting from existing asterisk
installations.
Th...
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 b...
2015 Jan 29
0
any valid up-to-date info about Kamailio-Asterisk integration ?
On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv at mail.ru> wrote:
> 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 an...
2009 Jul 27
2
Asterisk and Kamailio NAT problem
Hello
Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is
behind NAT.
X-Lite and SNOM phones behind NAT work fine.
But when I try to connect with an Asterisk behind NAT, the Asterisk
client doesn't receive sound.
I already tried in 2 different NATs, with no firewalls.
This is my Asterisk config:
[kamailio]
type=peer
hos...
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as Kamailio. The version is 11.10.2. With
Kamailio I use rtpengine, which affects SDP descriptions when 488 response
is received.
My goal is to enable two web...
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...
2016 Jul 05
2
OpenSIPS or Kamailio based fronting for Asterisk?
Hello,
I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so
far my biggest issue is the complete lack of quick-start-like documentation
for either. Is there any place I can get a very simple HA configuration
(telling me where the config files are, for starters, is a good thing) for
OpenSIPS or Kamailio with the following features:
(a) Support an arb...
2013 Jun 28
1
Asterisk behind NAT and Kamailio --> Internal IP in SDP and not "externip"
Hi,
We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4
NAT works very good with the externip/localnet-setting when we are
connected directly to our teleco. But when I try to use NAT and put them
behind our Kamailio something interesting happens: The media-address in
the SDP is the internal ip and not the external.
This is the setup:
Teleco <-> Kamailio <-> Asterisk
SIP --> 1.2.3.4
10.0.0.1 --> 10.0.0.2
externip=1.2.3.5
lo...
2013 Feb 11
0
Possible Security issue with Kamailio - Asterisk Realtime integration
Hi
I have an installation based on Daniel-Constantin Mierla's excellent
Kamailio 3.3 / Asterisk 10 Realtime document (
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
but have come across an issue which is a potential problem.
In this installation all SIP clients register with Kamailio, and the
registrations are forwarded to Asterisk. This means th...
2013 May 14
1
Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial
Hello,
I spent a bit of time to update my Kamailio-Asterisk realtime tutorial
to latest stable versions in both sides. The tutorial is available at:
-
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
I tried to use default names for asterisk database tables, where the
structure was not changed, and different names fo...
2014 Apr 24
1
Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing b...
2012 Aug 03
1
asterisk realtime database structure
...https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure
https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage
I have several table structures from the Asterisk 1.6, I dug for them in
the code or found on the web when I wrote the tutorial about integration
with Kamailio 3.1
(http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb),
but hopefully now it is an easy way to get the db structure.
Thanks,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced...
2015 Mar 10
1
PJSIP and Kamailio without registration
OK, it stopped working.
It turns out the transport and endpoints in PJSIP are ok. I can send an
invite from my unregistered snom phone and I can see some activity in the
CLI.
However, when I dial from my snom to Kamailio and have it pass the message
to asterisk, PJSIP seems to ignore the sip messages even though they are
there.
Is there something wrong in the invite that I'm missing?
U 2015/03/10 22:47:43.539208 [kamailio public ip]:5060 -> [asterisk public
ip]:5061
INVITE sip:1000 at somedomain.com;user=p...