Thanks Daniel, I wasn't starting rtpproxy properly. So the initial error is gone. But in my script when the rtpproxy_stream2uac is called i get the following log:
Dec 22 08:18:44 vps daemon.err /usr/sbin/kamailio[7901]: ERROR: rtpproxy [rtpproxy.c:1581]: script error -no valid set selected
Dec 22 08:18:44 vps daemon.err /usr/sbin/kamailio[7901]: ERROR: rtpproxy [rtpproxy_stream.c:113]: no available proxies
I have given the following commands at the beginning of the call: set_rtp_proxy_set("0");
rtpproxy_manage();
And when hold is pressed I've called stream2uac. Where am I going wrong? Could you also tell if rtpproxy_stream2uac will be able to play .wav files directly?
Thanks for your help.
Gautam
On Thu, Dec 22, 2011 at 7:38 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Can you give the output of:
ps auxw | grep -i rtpproxy
That will show if rtpproxy is running and what is its control
socket.
Cheers,
Daniel
On 12/21/11 11:25 PM, Gautam Batra wrote:
I'm not able to set up the rtp proxy module. I have
entered the following:
Where X.Y.Z.W is the IP address of my machine (same as that of my
SIP server). But the log shows the following errors:
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1503]: can't send command to a RTP proxy
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1538]: proxy <udp:X.Y.Z.W:22222> does
not respond, disable it
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1432]: support for RTP proxy
<udp:X.Y.Z.W:22222> has been disabled temporarily
Could anyone tell what I'm doing wrong? I tried to run rtpproxy
separately on the given port before starting kamailio (rtpproxy -s
udp:X.Y.Z.W:22222), but it didn't work.
I am using Freeswitch as an SBC behind Kamailio, and my
external calls are routed via freeswitch. In those calls the
music on hold works as it is handled by freeswitch. Ideally I
would like to somehow redirect when a call is put on hold to
the MOH extension. The other option is by using rtpproxy. I
could not find any documentation on rtpproxy and would really
appreciate it if someone could lead me to it or give me a
brief overview on how to go about using rtpproxy_stream2uac to
play music whenever a call is put on hold.
On Wed, Dec 21, 2011 at 4:50 AM,
Daniel-Constantin Mierla <miconda@gmail.com>
wrote:
Hello,
On 12/21/11 7:49 AM, Olle E. Johansson wrote:
20 dec 2011 kl. 22:40 skrev Gautam Batra:
Hi,
Thanks for your replies. Is it possible to play
an audio file in the case of a re-invite
directly from kamailio instead of freeswitch by
using rtpproxy_stream2uac() or something
similar?
Kamailioi is still a proxy and from the endpoint
point of view is not involved in the media plane.
If you managed to do that many
endpoints would ignore the packets or see them as
a DOS attack attempt. Other endpoints might just
play them.
In later releases of Asterisk, we lock to the IP
address of the peer and would ignore these.
Asterisk used to send music-on-hold
like this before, but we considered it a security
issue and started reinviting to make Asterisk
involved in the call again to play
music on hold. Asterisk can do that, because it's
a b2bua and is an endpoint in the call. Kamailio
can't initiate a reinvite in the
call.
indeed, kamailio cannot initiate re-invites. You can
play an audio file via rtpproxy and
rtpproxy_stream2uac() if you use rtpproxy relaying
from the beginning of the call. Otherwise, use a sip
b2bua which does signaling only until you need to play
audio and do re-invites so it gets in media path.
Besides Asterisk or FreeSWITCH, a lightweight b2bua
that probably offers such functionality is sems (sip
express media server) -- I CC-ed Stefan, he can
confirm and even give some leads of how to do it.
Cheers,
Daniel
/O
Gautam
On Mon, Dec 12, 2011 at 4:50 AM, Olle E.
Johansson<oej@edvina.net>
wrote:
12 dec 2011 kl. 10:45 skrev Daniel-Constantin
Mierla:
Hello,
On 12/9/11 9:04 PM, Gautam Batra wrote:
Hello,
I have a kamailio sip proxy server with
freeswitch acting as SBC. I want to redirect
the call to freeswitch when hold is pressed
so that i can play music on hold. I tried
this by using rewritehostport in case of a
re-invite, but the call drops in that case.
Could someone please help me with this?
it is not possible to redirect established
calls (it breaks the RFC3261), you have to
route the call through freeswitch from its
start. Perhaps you can use freeswitch without
relaying the media in first place and when you
have on hold, you set media patch to go
through freeswitch.
The only solution is having FreeSwitch send an
invite with replaces to grab the call. The
question is how to get it back.