Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help.
--Vivek On 11/10/07, Ryan Newington <[EMAIL PROTECTED]> wrote: > > Hi Luki, > > Thanks for your advice. I've checked the firewall and it is set to allow > all incoming traffic. I changed the media port range as well with no > success. > > Some calls work fine. This is the configuration that doesn't work. The RTP > traffic passes along the chain fine, but the Asterisk server doesn't do > anything with the packets it gets from the near-end SIP phone and the media > gateway. > > SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone > > An asterisk internal call will work fine. Eg; > > SIP Phone <-> Asterisk <-> SIP Phone > > Regards > > Ryan > > > > -----Original Message----- > From: [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] On Behalf Of Luki > Sent: Sunday, 11 November 2007 12:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] RTP traffic not being forwarded > > > When using 'rtp debug' on the asterisk console, it shows that it is > > receiving traffic from one endpoint, but not the other. A wireshark > trace > > reveals it is actually receiving traffic from both ends. > > Sounds like a firewall issue. Wireshark shows what's "on the wire", > i.e. before iptables. The packets are being dropped for whatever > reason and never reach the asterisk process. Check your iptables and > RTP port range, and perhaps try changing it. > > Luki > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > 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-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users