Displaying 20 results from an estimated 1000 matches similar to: "Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available"
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 but for example realtime load
sipusers name 660 shows the
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
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
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 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 communicate with each other.
I have a Snom phone accessing Kamailio via its
2014 Aug 11
1
Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk changes them in the sdp bodies along the way. I'm using Asterisk
11.11.0.
Is there a way to
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
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.
The other
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
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
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
host=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
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
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
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
2012 Aug 03
1
asterisk realtime database structure
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk version or a web resource/file containing
the sql scripts. Hope I haven't missed obvious things, I had no luck
searching on the web, in the wiki I found few pages with bits of sql or
table structures, like:
2009 Jul 06
1
Asterisk + kamaili MWI(Message waiting Indication)
hello,
Does anyone know about setup Message wait indication between asterisk and
kamailio
my phone are registered on kamailio and voicemail leaves on asterisk
server.
how do i notify to kamailio that 1 message is leaved for you on your
mailbox.
and also i tried all script listed in voip-info.org.
any one know any working method or anybody have some type of setup which may
help me
any help
2014 Jul 15
1
Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Hello all,
I have an Asterisk installation with Kamailio using realtime integration. I
have gotten my clients to register, but there is something odd about the
sip message flow with some of my clients. My clients are Zoiper and
Asterisk is 11.10.2.
When I set 'Subscribe to MWI' value to 'both', after a normal, successful
registration Asterisk begins to send REGISTER messages to
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 for those that are a bit
customized, in order to
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 arbitrarily large number of