Mute call from messagenet to sip hard phone

Asterisk Asterisk General
1

I all,
I have installed Asterisk 1.8.11 and i put on a simple configuration.
I installed and registered 4 SIP (hard/soft phone) account and 1 messagenet.

below my “sip show peers”

localhosrvasterisk*CLI> sip show peers
Name/username              Host                     Dyn Forcerport ACL Port     Status     
700/700                    *.*.*.11                              D   N             5060     OK (4 ms)  
701/701                    *.*.*.16                              D   N             5060     OK (40 ms) 
702/702                    *.*.*.24                              D   N             5060     OK (4 ms)  
703/703                    *.*.*.12                              D   N             5060     OK (58 ms) 
messagenet/5******    212.97.59.76                          N             5060     OK (13 ms) 
5 sip peers [Monitored: 5 online, 0 offline Unmonitored: 0 online, 0 offline]

more “sip show registry”

localhosrvasterisk*CLI> sip show registry 
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
sip.messagenet.it:5060                  N      5******            105 Registered           Thu, 28 Jun 2012 21:02:59
1 SIP registrations.

below the relation between the extensions and the type of hardware/software used

700 -> Grandstream BT100
701 -> Elmeg Snom 190
702 -> Ekiga (Softphone on Kubuntu)
703 -> V-IP400P

When i make a call from 700 to 701,702 or 703 (or i make a call from 701 to 700,702 or 703, or i i make a call from 702 to 700,701 or 703) Hopefully, instead, when i make a call from my cellular to my messagenet number that is redirected to 700 or 701 or 703 both terminals (my cellulare and one SIP phone) are dumb.
Only when i make a call from my cellular to my messagenet number that is redirected to 702 hopefully.

2

Help me please!!!

3

I don’t understand this sentence at all.

You need to provide the dialplan and sip.conf contents. You also need to provide a trace of the SIP and SDP exchanges.

You need to make it clear which addresses are routable and which are not, even if you obfuscate them.

Generally, lack of audio indicates a firewall or NAT configuration error.

This is also the wrong forum for support questions, so some people may have missed it.