Hi You should uncheck the "anonymous call" & "allow guest" in asterisk
On Sat, Apr 29, 2017 at 2:15 PM, samaneh ebrahimi <[email protected]> wrote: > Hi , > For setup sip authentication between Cisco router and FreePBx , configs is > : > > [image: Inline image 1] > *PBX - SIP Trunk* > PEER Details > > fromdomain=40.0.0.100 > host=40.0.0.100 > qualify=yes > context=from-trunk > type=peer > username=asterisk > secret=password > nat=no > allow=alaw > faxdetect=yes > insecure=very,port,invite > > *cisco router* > > voice service voip > allow-connections sip to sip > ! > interface FastEthernet0/0 > ip address 40.0.0.100 255.255.255.0 > ! > interface FastEthernet0/1 > ip address 192.168.0.100 255.255.255.0 > ! > dial-peer voice 1 voip > destination-pattern 1.+ > session protocol sipv2 > session target ipv4:40.0.0.1 > codec g711alaw > ! > dial-peer voice 2 voip > destination-pattern 2.+ > session protocol sipv2 > session target ipv4:192.168.0.101 > codec g711alaw > ! > sip-ua > authentication username asterisk password 01030717481C091D25 > retry invite 3 > retry response 3 > retry bye 3 > retry cancel 3 > sip-server ipv4:40.0.0.1 > > by this config, calls is established and i expect when remove > authentication from FreePBX or router , calls Not contacted . but it is > established . > > > _______________________________________________ > cisco-voip mailing list > [email protected] > https://puck.nether.net/mailman/listinfo/cisco-voip > >
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