Le 30/05/2012 14:44, Matthew J. Roth a écrit :
Considering that you made progress on your initial problem by setting "nat=force_rport" (resulting in connected calls with no audio) and now you're mentioning the use of "externaddr", I'd recommend a very careful reading of the "NAT SUPPORT" section of sip.conf.sample in the configs directory of the Asterisk source tree.
I did read all those documentation, belive me. Also keep in mind that I *ONLY* face this problem with this provider, people using voipbuster or sipdiscount should have the same problem.
Concerning externaddr, this test server -dedicated to asterisk- being running in VM since ages, I never would suspect a NAT issue! Especially if previous 1.4 and 1.6 version are running smoothly ...
In Asterisk 1.8, there is a new configuration option named "media_address" which may be of particular interest.
media_address seems not an option, can be set only in general not per peer.
This is confusing because your first email said you had "nat=no" in your working 1.6.24 setup, but everything you're saying indicates a NAT problem to me.
Again, 1.6 version is perfectly working with this setup and conf files, and before 1.4 was too. And those both asterisk versions with *this* provider.
. A diagram showing all network elements between your Asterisk server and the remote host would be helpful.
Phone registration:phone (Snom320 and GS GXV3175) -> firewall1 (linux router) -> Internet -> firewall2 (linux router) -> VM -> phone account
Call:phone account -> Out of VM -> firewall2 (linux router) -> Internet -> Peer IP -> ???
To avoid further confusion, please include full and unaltered logs, SIP traces, and configurations in future posts.
During the time you and Andres replied to my post ;-) I got the same idea then him; and guess what, it's working! So problem is Asterisk 1.8/10 in VM _only_ this provider(s) which are all Dellmont services.
Can someone confirm the problem? Question is now, who is faulty? Should I open a bug? Thanks for your time and support. -- Daniel -- _____________________________________________________________________ -- 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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