[asterisk-users] Windstream SIP Trunk settings

[James Cass]

Thanks everyone, all sound advice.  Still can't even get the calls to show
up on the console at all - I suspect the issue is on the WS side, as I'm
not having any issues with other carriers with similar settings.

Thanks again.

James Cass <http://goog_987864563>
jcass78 at gmail.com


On Mon, Feb 22, 2016 at 9:06 AM, Mark Wiater <mark.wiater at greybeam.com>
wrote:

> In my case, username is the BTN. I also set the fromdomain to be the sbc
> that I'm registering with. Externip might help also?
>
> [paetec]
> host=10.250.0.5
> username=btn
> fromdomain=10.250.0.5
> dtmfmode=rfc2833
> externip=10.255.0.2
>
> I've used these settings on both registering and non-registering trunks,
> connecting to both the Broadworks and Plexus platforms in Windstream.
> Though all of my asterisk versions have been 1.8.x
>
> Mark
>
>
> On 2/22/2016 8:20 AM, James Cass wrote:
>
> Does anyone on this list use Windstream as a SIP trunk provider?
>
> If so, would you mind sharing your peer settings?
>
> I'm using asterisk 13.7.2 and can't seem to get the inbound working
> correctly (using registration).  Outbound is fine, but they are seeing an
> authentication error on their end.
>
> Here are my inbound peer settings:
>
> username=<accountnumber>
> secret=<secret>
> host=<host address>
> type=peer
> fromuser=<accountnumber>
> context=from-trunk
> dtmfmode=auto
> canreinvite=no
> qualify=yes
> insecure=port,invite
>
> register string:   <accountnumber>:<secret>@<host address>:5060
>
> Thanks in advance,
>
>
>
>
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