[cisco-voip] sip-authentication by cisco router and Asterisk

[Esmaeel Saberi]

Hi
You should uncheck the "anonymous call" & "allow guest" in asterisk

On Sat, Apr 29, 2017 at 2:15 PM, samaneh ebrahimi <esamaneh.94 at gmail.com>
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 .
>
>
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>
>
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