VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
jcass78 at gmail.com Guest
|
Posted: Mon Feb 22, 2016 8:21 am Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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: Â Â <accountnumber>:<secret>@<host address>:5060
Thanks in advance, |
|
Back to top |
|
|
mailinglist at linuxis... Guest
|
Posted: Mon Feb 22, 2016 8:32 am Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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
--
_____________________________________________________________________
-- 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 |
|
Back to top |
|
|
mark.wiater at greybea... Guest
|
Posted: Mon Feb 22, 2016 9:07 am Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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,
|
|
|
Back to top |
|
|
jcass78 at gmail.com Guest
|
Posted: Tue Feb 23, 2016 7:38 am Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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: Â Â <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
|
|
|
Back to top |
|
|
decipher.hk at gmail.com Guest
|
Posted: Tue Feb 23, 2016 12:20 pm Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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 RamÃrez 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 |
|
Back to top |
|
|
jcass78 at gmail.com Guest
|
Posted: Thu Mar 03, 2016 2:25 pm Post subject: [asterisk-users] Windstream SIP Trunk settings |
|
|
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 -Â
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 RamÃrez 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 RamÃrez 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
|
|
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|