To Answer Some of You Questions: Please not that I replace the true domain wtih "example", and the true ip for the remote UA with "public-ip". Nothing against no one here, just don't know who else would read this email in the future!!!
PS: The public IP of the remote UA is correct. SIP Show Peers: Name/username Host Dyn Forcerport ACL Port Status Realtime 1002/[email protected]. 192.168.2.13 N 5060 UNKNOWN Cached RT 1003/[email protected]. -public-ip- D N 5060 OK (86 ms) Cached RT Peers look registered correctly. This has now become a sip proxy issue :S. Thank you so much for your time guys!!! N. On 1/3/13, Nick Khamis <[email protected]> wrote: > 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
