There is a sip number and an asterisk server in office 1, and there is office 2 in another network 20 km away, how to make site to site between networks so that phones in office 2 work?

Question

Asterisk

The problem is that the provider blocks the phone's sip port and the voice disappears through vpn. what to do ??

Answer 1

2 cheapest mikrotiks, gre tunnel, routing, set up and there will be happiness.

Answer 2

Tune. First, figure out why the voice disappears. Most often this is due to the peculiarities of sip. Since a different protocol is used for voice, rtp, and only a connection is established via sip.

Answer 3

I raised DMVPN between branches on tsiska 1921, everything worked stably.
And what's the problem with replacing the sip port of the phone? In the asterisk and in phones, they seem to be changing.

Answer 4

vpn ipsec tunnel, for example
, you can also quarrel with the provider

Answer 5

If the provider blocks exactly the traffic sent to port 5060, and not something else, you can do it simply:
In the sending router, configure the outgoing traffic to change the SIP port 5060 to some left number, for example 1010.
And in the recipient router we set up a reverse replacement for 5060 for incoming traffic from port 1010 and, at the same time, redirect traffic to the desired local IP where the PBX is spinning.