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[asterisk-users] Windstream SIP Trunk settings


 
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jcass78 at gmail.com
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PostPosted: Mon Feb 22, 2016 8:21 am    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

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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mailinglist at linuxis...
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PostPosted: Mon Feb 22, 2016 8:32 am    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote:

Quote:
register string: <accountnumber>:<secret>@<host address>:5060

Try:

register => 5551231234:SECRET@sipdomain.com/5551231234


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mark.wiater at greybea...
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PostPosted: Mon Feb 22, 2016 9:07 am    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

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:

Quote:
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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jcass78 at gmail.com
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PostPosted: Tue Feb 23, 2016 7:38 am    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

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.

jcass78@gmail.com (jcass78@gmail.com)








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

Quote:
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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decipher.hk at gmail.com
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PostPosted: Tue Feb 23, 2016 12:20 pm    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

February 23 2016 9:37 AM, "James Cass" <jcass78@gmail.com> wrote:
Quote:
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 Ramírez Norambuena
http://www.rodrigoramirez.com

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_____________________________________________________________________
-- 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

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jcass78 at gmail.com
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PostPosted: Thu Mar 03, 2016 2:25 pm    Post subject: [asterisk-users] Windstream SIP Trunk settings Reply with quote

Here's what I ultimately got to work (in case it helps someone):


Name your trunk
Enter your outgoing CID
Under Outgoing settings-
Trunk name - whatever you choose to name it
PEER Details-
host=IP address of SIP gateway
type=friend
context=from-trunk
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite


Incoming settings - 

none
Registration string -
username:password@xxx.xxx.xxx.xxx


jcass78@gmail.com (jcass78@gmail.com)








On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena <decipher.hk@gmail.com (decipher.hk@gmail.com)> wrote:
Quote:
February 23 2016 9:37 AM, "James Cass" <jcass78@gmail.com (jcass78@gmail.com)> wrote:
Quote:
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 Ramírez Norambuena
http://www.rodrigoramirez.com

--
_____________________________________________________________________
-- 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

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