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

[Brian West]

Was this not helpful? http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

/b

On Nov 6, 2008, at 8:23 AM, Peter P GMX wrote:

> 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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