Yes, it requires a very particular configuration. With asterisk you in
sip.conf you need
externaddr=you.public.ip
This is used in SIP registrations by Asterisk. You should also be
comfortable with debugging the SIP traffic to make sure all IP's are set
properly. For asterisk:
sip set debug on
Look for REGISTER messages.
Another amazing tool for this is sipgrep (
https://github.com/sipcapture/sipgrep). It's like ngrep but for SIP.
Benjamin Fitzgerald
LETS Corporation
(925) 566-5600 ext. 25
ben(a)letscorp.us
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On Fri, Aug 11, 2017 at 3:29 AM, <vs(a)cell.tel> wrote:
Hi!
Could anyone advice me or provide an example of Kamailio config to solve
this case:
Current Kamailio+Asterisk schema:
(7XXX Users) <-> Asterisk <-> NAT <-> Internet <-> Kamailio
<-> (4XXX
Users)
7XXX Users registers on Asterisk server, and 4XXX Users registers on
Kamailio.
In Addition, Asterisk registers on Kamailio with user 4444, and it is
possible to make calls from 7XXX Users (Asterisk) to 4XXX Users (Kamailio)
through 4444 "trunk".
Also, while Kamailio users (4XXX) call to 4444 number, call successfully
goes to Asterisk.
How its possible to route all 7XXX destined calls from Kamailio users to
Asterisk, while Asterisk is behind NAT???
Config examples, i found, were writen only for situation with fixed
asterisk IP and PORT, without nat ( asterisk.bindip = "" ; asterisk.bindport
= "")
Thanks in advance
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