I have remote access to the server so I checked the canreinvite .. they are all set to no. I can't try the call from here, I will get back to you.
Gary Baribault On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote: > > Do you agree something is blocking the audio in one direction? Can you > do a 'rtp debug' and then initiate a SIP call and see if there is two > way audio traffic. Also make sure these extensions have 'canreinvite=no'. > > Zeeshan A Zakaria > > -- > Sent from my Android phone with K-9 Mail. > >> On 2010-06-01 7:02 PM, "Gary Baribault" <[email protected] >> <mailto:[email protected]>> wrote: >> >> As I stated, the incoming calls are on TDM DS0s connected to the >> Digium card, and the extensions are on the same local network as the >> Asterisk server. There is currently no NAT anywhere. >> >> Gary Baribault >> >> >> >> On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: >> > >> > Output of 'iptables -L -n' would also be helpfu... >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users
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