Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: beckynet on February 07, 2010, 05:31:36 PM
-
In Sail 2.4.7 and precedent 2.4 version the panel to create or manage new carrier has disappeared...
I want to create one for my sip provider 3starsnet.com
Thanks a lot
Olivier Beeckmans
-
type pls why General SIP is not enough sufficient.
i think you can set all parameters you need.
if there is something extraordinary I'll try to give db command to add it to proper datebase
-
HI Olivier
We've removed it in 2.4. As Maciej points out above, "General SIP" should cover pretty much any SIP requirement.
Best
S
-
Hello,
I've used General Sip, but I've some trouble.
I can use This VOIP trunk to call external phone. But when someone try to contact me I receive nothing. And there are none activity in asterisk -rvvvvv when I try to call my VOIP trunk with my Mobile Phone.
On the same server, before I use Dahdi and the last SAIL-2.4.x this line can send and receive call.
Here for info you can find support from 3Starsnet for Asterisk
http://www.3starsnet.com/support/pdf/asterisk.pdf
Thanks Olivier Beeckmans
-
Just create a General SIP trunk which looks like the peer entry in their example and fill out the registration details. You may also need to create a DiD trunk to receive the inbound calls.
I've put a couple of screen grabs up for you to look at...
http://sarkpbx.com/images/3stars.jpg
http://sarkpbx.com/images/3stars2.jpg
Create as I've done in the first image and then go to edit the trunk and add the missing lines to the left hand window from the 3stars peer entry example. Don't use any of their entries from the general section because they are a bit silly.
You should be good to go.
Kind Regards
S