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, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160222/809fdd03/attachment.html>
On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote:> register string: <accountnumber>:<secret>@<host address>:5060Try: register => 5551231234:SECRET at sipdomain.com/5551231234
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, > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160222/6afb81cf/attachment.html>
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, > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160223/e15589ff/attachment.html>
Rodrigo RamÃrez Norambuena
2016-Feb-23 17:20 UTC
[asterisk-users] Windstream SIP Trunk settings
February 23 2016 9:37 AM, "James Cass" <jcass78 at gmail.com> wrote:> 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.You can debug SIP to detect the problem. May be exists some cause tell you more information in the trace SIP. -- Rodrigo Ram?rez Norambuena http://www.rodrigoramirez.com