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
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