Re: [SR-Users] Music on hold with freeswitch as SBC

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 &lt;miconda(a)gmail.com > 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: > > loadmodule "rtpproxy.so" > modparam ("rtpproxy", "rtpproxy_sock", "udp:X.Y.Z.W:22222"); > > 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. > > > > On Wed, Dec 21, 2011 at 2:36 PM, Gautam Batra &lt;gautambatra24(a)gmail.com&gt;wrote;wrote: > >> 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(a)gmail.com>> >>> 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&lt;oej(a)edvina.net&gt; >>>>> 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. >>>>> >>>>> /O >>>>> >>>>> >>>>> --- >>>> * Olle E Johansson - oej(a)edvina.net >>>> * Cell phone +46 70 593 68 51 <%2B46%2070%20593%2068%2051>, Office +46 >>>> 8 96 40 20 <%2B46%208%2096%2040%2020>, Sweden >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.com >>> http://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda > >