Oooops yes of course 10004-10007!! Simple math does not come easy anymore... Anyhow, I singled out Opensips and I have two way audio form UA(local) -> UA(remote) but not from UA -> Siptrunk. That being said maybe a small diagram of the architecture. Please don't laugh!!! :) I know having a block of static IPs would make like easier however....
UA (Remote) -> Router (Remote) -> Internet -> Router (Local) -> OpenSIPS+RTPProxy -> Asterisk Port forwarding (Remote): 5060, and 10000-50000 to UA Port Forwarding (Local): 5060. and 10000-50000 to OpenSIPS) "No Audio" Port Forwarding (Local): 5060. and 10000-50000 directly to Asterisk "Two Way Audio" Cheers Guys! Nick On 1/3/13, Danny Nicholas <[email protected]> wrote: > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Jason Parker > Sent: Thursday, January 03, 2013 2:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Moving User Agent To Remote Location > > On 01/03/2013 02:23 PM, Markus Weiler wrote: >> Am 03.01.2013 21:21, schrieb Nick Khamis: >>> Oh that's so smart!!! So, if I did not misunderstand you, for this >>> one call, have: >>> rtpstart=10004 >>> rtpend=1008 > > The rtpend should be 10008 and rtpstart should be 10005. A SIP call in > Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels > for audio. AFAIK the odd channel is send and the even channel is receive > (smarter folks than me like Tzafir can give you the specifics; this was > covered at least twice in 2012 threads). If you open 5060 on your > NAT/firewall, but open no RTP channels, you will establish a call with no > sound. > > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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
