Here ya go. Its a very basic configuration. We basically just use this server as a sip/media proxy for some call center agents so we can record the audio from a central location. The calls are generated from a shoretel system which sends the SIP calls to the asterisk box then the asterisk box sends everything on to the carrier sip address. The agents are around 30 at a time. They do outbound calls and then conferencing. Thanks. ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=10000 rtpend=20000 ; ; Whether to enable or disable UDP checksums on RTP traffic ; ;rtpchecksums=no ; ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 ; rtcpinterval = 5000 ; Milliseconds between rtcp reports ;(min 500, max 60000, default 5000) ; ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; disabled by default. ; strictrtp=yes On Fri, Sep 16, 2011 at 1:05 AM, Alex Balashov <abalashov at evaristesys.com> wrote:
Just out of curiosity, what's your /etc/asterisk/rtp.conf say?
-- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops