Testing a 2.0.5 install and cannot get it to dial out SIP

[brown078]

Hello all,
I am working through a 2.0.5 stable install with asterisk 1.2.30 on debian.

I have SIP/teliax registered and terminating outbound calls through the softphone.

However, when I log in to a manual dial campaign with an phone set with "SIP/teliax" as the outbound group, it still seems to want to send through "Zap/g0" no matter whats changed.

Is there a file I could modify this on?

Thanks and congrats on the new stable, exciting changes!

[mflorell]

Sounds like a dialplan issue. Is this an upgrade from an earlier version?

How are you configuring your trunks?

[brown078]

Thanks for your response Matt,
I am trying to use the "providers" functionality in the admin.
I wanted to configure the box in accordance with the "template" style
This worked just fine on an outbound call. Its when we log into the campaign.

Thanks

[mflorell]

What is the dial prefix that you are using to dial out of your campaigns?

What does your carrier entry dialplan section look like?

[codehaxor]

just replace

TRUNK and TRUNKX values to make it more easier