Displaying 20 results from an estimated 4000 matches similar to: "Split SIP and RTP to different IP addr"
2012 Jun 22
2
a2billing
hello,
I just installed a2billing, I did all the config, at least I guess ..
but I still can not integrate sip accounts that I had created (with sip.conf
) in a2billing to apply their billing ..
could someone tell me how to make the junction between asterisk and
a2billing??
I also noticed that the file
additional_a2billing_sip.conf : was always empty ...
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2011 Dec 29
2
Interesting attack tonight & fail2ban them
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example:
[2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension
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
2014 Apr 14
1
Webrtc and adventures with Asterisk 11
Hi,
I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 +
opus/vb8 codec patch. This is interesting technology and I try to find
out how to connect all the moving parts.
Firefox:
Neither sipml5 or jssip works with calls to asterisk, audio/video
doesn't matter.
WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream
without encryption details: audio 35684
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
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
2012 Oct 05
2
SendFAX - multi-page TIFF
Hi,
Does anyone had the problem of asterisk SendFax + spandsp sending only
the first page of a multi-page TIFF file?
Seams to be related to spandsp ECM config.
Any thoughts about it?
Thanks,
Gabriel
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2011 Jul 04
4
stream rtp from asterisk
Hi!
Anybody familiar with streaming rtp from asterisk. Preferably with the
xorcom asterisk patch which streams rtp from asterisk to oreka audio
server. Any ideas will do just fine though!
Regards / Marcus
2011 Dec 15
1
Wrong call information on B leg
Greetings.
I have next feature in features.conf :
send =>
*9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl
What it does is parsing
CALLERID and DNID from AGI input, performing some actions in MySQL with
these values, and then running application for peer (for example,
PlayBack)
Sounds simple, and it really is. When my user is receiving a
call (we are the B leg) and presses *9, everything
2011 Jun 09
1
Fwd: Re: ControlPlayback's options
Humm... Seems like my message didn't make it. Here we go again..
/Johan
-------- Original Message --------
Subject: Re: [asterisk-users] ControlPlayback's options
Date: Sun, 05 Jun 2011 22:19:18 +0200
From: Johan Wilfer <lists at jttech.se>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
On 2011-06-05 19:54, virendra
2013 Oct 07
1
IAX and Variables
Hi
a new small question ;=)
We have two Asterisk, connected in IAX2.
On the first, in dialplan, we have:
exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
we sent into the IAXVAR "ACCOUNTID" the accountcode.
On the second, in dialplan, we have:
exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
That's work, the second server get the variable.
I
2013 Jul 26
1
Random dead calls
Hi,
Am getting dead or silence calls at sometimes for my agents, when I checked
my CDR the caller-id shows my vendor's name and some shows as real customer
name.
When I call back again the real customer's number its reaching, the
answering machine owned by customer.
I have a confusion, or how to find out are these numbers are from any auto
dialer or from real customers.
Thanks.
2012 Aug 02
1
Originate call from cli does not work for SIP line...
I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to
dial from the CLI using the originate command or use an AMI connection
to originate a call I get the following error:
originate SIP/protel-out/0445540881644 application playback tt-monkeys
WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received
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
2009 Apr 02
1
SIP vs RTP destination IP
Is it possible to have asterisk override the connection information embedded
in a SIP 200 packet with the registration information? I have multihomed
machines with softphones and they register just fine and sip works fine, but
the RTP packets get sent to the ip from the SIP connection information and
the softphones are sending the wrong ip. I can't find an option in the
softphone to change ip
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 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