Hi Stefaan,
the documentation on this feature is indeed very poor.
I've set it up once and never changed the lines of code again (and i
also only created the announcements once).
Carsten
2011/7/12 <s.maertens(a)telenet.be>be>:
Hi Carsten,
Thank you for your feedback.
I have been able to get some more information and improvement by reading the sources of
rtpproxy.
The calls are going over the RTPProsxy and the sound is now indeed being streamed by
RTPProxy (only about 3 times too fast :) )
Maybe the reason is that I used another box to encode the files message.wav with makeann
(from RTPProxy sources) to message.0 and message.8 or there is something else wrong with
the format of the file.
I am still trying to solve this last issue and will then write a summary of what I have
done or used as config to make this work.
imvho the documentation of kamailio and the rtpproxy module is very clear , rtpproxy
itself and especially the makeann program is poorly documented.
Best regards
Stefaan
----- Originele e-mail -----
Van: "Carsten Bock" <carsten(a)ng-voice.com>
Aan: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing
List" <sr-users(a)lists.sip-router.org>
Verzonden: Dinsdag 12 juli 2011 16:24:51 GMT +01:00 Amsterdam / Berlijn / Bern / Rome /
Stockholm / Wenen
Onderwerp: Re: [SR-Users] rtpproxy_stream2uac
Hi,
did you send the calls over the RTPProxy in the first place? If the
calls are not going through the RTPProxy, the calls will not work...
Carsten
2011/7/8 <s.maertens(a)telenet.be>be>:
Hi,
I am trying to get kamailio (or more specific rtpproxy) to play a wav file
when a call is setup
In kamailio.cfg I have added the command in the onreply_route
rtpproxy_startrecording();
rtpproxy_stream2uas("/var/log/rtpproxy/message","-1");
I am sure the command is executed and given from kamailio to rtpproxy
because in syslog I see following output (using dbug output of rtpproxy)
rtpproxy[2648]: DBUG:handle_command: received command "P-1
024a73d95bbe016014de647e700 /var/log/rtpproxy/behappy session 487095817;1
024a73d95bbe015f14de647e700;1"
rtpproxy[2648]: DBUG:doreply: sending reply "E6#012"
rtpproxy[2648]: DBUG:handle_command: received command "R
024a73d95bbe016014de647e700 487095817 024a73d95bbe015f14de647e700"
rtpproxy[2648]: DBUG:doreply: sending reply "0#012"
The "R" command is the startrecording command and that is indeed working
perfect. (recording to /var/log/rtpproxy)
The "P" command is the "Playing or stream2uas" command. Seems that
the
message rtpproxy gets differs a bit from the one I found in the rtpproxy
manual
from the manual :
P[args] callid play_name codecs from_tag to_tag
Direction of the playback is defined by the order of the from_tag and to_tag
parameters.
R callid from_tag to_tag
I have used the command " makeann behappy.wav behappy" to convert a
wavfile. It gave me a .0 and .a file which I have placed in the
/var/log/rtpproxy directory (makeann comes with the rtpproxy source and is
rather a bit shy with information to say the least ;) )
There is not too much information or examples found about this so I was
hoping maybe somebody on this list could help me in finding out why i'm not
getting more information in syslog and why i'm not hearing the audiofile.
Best regards
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--
Carsten Bock
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Schomburgstr. 80
22767 Hamburg
Germany
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
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_______________________________________________
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
_______________________________________________
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
mailto:carsten@ng-voice.com
Schomburgstr. 80
22767 Hamburg
Germany
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
~~~~~
Checkout SIP-Provider CE: