forum.newfies-dialer.org

We have a system running about 100-200 channels broadcasting about 50,000 calls a day.

We are going to test press 1 today. We have a newfies dialer and elastix(asterisk) connected via sip trunk. If somebody presses 1 then the call goes through the SIP trunk to elastix via a SIP URI

Newfies = 192.168.1.4
Elastix = 192.168.1.2


Newfies SIP trunk config to elastix:

<include>
<gateway name="192.168.1.2">
<!--/// account username *required* ///-->
<!--param name="Intelepeer" value="Intelepeer"/>-->
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<!--<param name="realm" value="asterlink.com"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--param name="from-user" value="your username provided by carrier"/>-->
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--param name="from-domain" value=""/-->
<!--/// account password *required* ///-->
<!--/// param name="password" value="your password supplied by carrier"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<!--<param name="proxy" value="asterlink.com"/>-->
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<param name="register" value="false"/>
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
</gateway>
</include>

Elastix SIP trunk to Newfies:

allow=all
type=friend
host=192.168.1.4
dtmfmode=rfc2833
context=from-internal
qualify=no
insecure=very


The Elastix call queue pilot number is 270

In newfies survey select call transfer and put in 270@192.168.1.2 in the SIP URI box. This works and I have tested it. We are going to go into test production with config today.