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[sambasivarao_vemula at...]

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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[faraz.khan at emergen.biz]

sambasivarao Vemula wrote:

[brian at freeswitch.org]

http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

/b

On Nov 5, 2008, at 12:31 AM, sambasivarao Vemula wrote:

[Prometheus001 at gmx.net]

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@asterisk.domain"/>
</condition>
</extension>

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

Best regards
Peter


sambasivarao Vemula schrieb:

[brian at freeswitch.org]

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: