[Freeswitch-users] Anybody tried with Trunk between asterisk and freeswitch...?

[Peter P GMX]

It's rather simple
- Setup a sip user on asterisk with username/password
- Setup a gateway in freeswitch with the asterisk user credentials (ip,
username, password of asterisk)
- Define a route in the dialplan (e.g. default.xml) to route certain
numbers to the asterisk gateway
e.g.
<extension name="Long Distance -Asterisk">
<condition field="destination_number" expression="^(0[2-9]\d{4,13})$">
<action application="set" data="effective_caller_id_number="/>
<action application="export" data="sip_secure_media=true"/>
<action application="bridge"
data="sofia/gateway/asterisk/$1 at asterisk.domain"/>
</condition>
</extension>

You should already be able to make outgoing calls via asterisk.

Best regards
Peter


sambasivarao Vemula schrieb:
>
>  
>
> Hi,
>
> Any body tried with Trunkig between freeswitch and asterisk .
>
> If any body tried and its working fine .
>
> Please share the details.
>
> Regards
>
> Samba
>
>  
>
>  
>
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