A similar issue happens to us. Make sure that, for inbound AND outbound calls rtp packets are reaching the other endpoint. If a NAT device(s) is between the endpoints make sure that the device NATs the traffic on BOTH ways (inbound AND outbound).
Regards On Saturday 27 September 2008 23:54:37 Philip Prindeville wrote: > I've got the following situation. I'm running Asterisk 1.4.18 on a > firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones > behind it. > > I'm peering SIP with a Coppercom switch sitting behind an SBC. > > On outbound calls, I get 2-way voice, no worries. > > On inbound calls, I get one-way voice (I can hear the caller but they > can't hear me). > > I've looked at tcpdumps of the RTP traffic, and the addresses and port > numbers correspond to what's in the SIP INVITE/OK messages (assuming > that they don't somehow get munged by NAT after tcpdump looks at them -- > there is no NAT device upstream of my Asterisk firewall). > > I'll look into using Record() or Monitor() to capture the phone call, > but if there's any conversion being done by codecs then that won't > eliminate the possibility that the code itself is misconfigured or buggy > and generating a bad stream on one of the legs... > > Anyone have an idea about how to best go about troubleshooting this? > > Thanks, > > -Philip > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
