There is no list that I’m aware of but then the 7000 series is a PBX/Feature Server on its own, so it wouldn’t really be tested with other PBX’s and feature servers. We do however have a list of SIP trunking providers that we work with ...
Could you perhaps be looking for information on a gateway device rather than our PBX product? Say for example TA900 or the NetVanta 6000 series?
Yes. That is exactly what I am looking for. The TA900 series and the
NetVanta 644. I have customers that like to do SIP Trunks and break that
out to a IP PBX, PRI, or POTS config. Also does the TA900 series and
NetVanta 644 prefer SIP Username and Password or IP based authentication?
On Fri, Apr 6, 2012 at 11:02 AM, racolvin <
You can find a list of tested & supported softswitches/feature servers in the AOS Feature Matrix - Product Feature Matrix , but here's is a listing: And yes, we use "SIP" Username "Identities" & Password Authentication.
Supported SIP Feature Servers Altigen Asterisk Broadsoft Metaswitch Nortel CS2000 Nortel MCS 5200 Shoretel Sonus ASX Sylantro
Thank you for your response. I tried to use SIP Username & Password and I
was getting a "403 response" when I changed it to IP based I was able to
make outbound calls. I am using Asterisk servers. Any thoughts?
On Mon, Apr 9, 2012 at 10:02 AM, burgermeister <
Below are configuration examples for how I typically setup username and passwords for voice users and trunks. Of course, you will need to swap phone numbers, auth-name, etc with your specific DIDs.
! SIP voice user (FXS)
voice user 1000
connect fxs 0/1
sip-identity 1000 T01 register auth-name "1000" password "1234"
! SIP trunk registration on behalf of a PRI or CAS trunk.
voice trunk T01 type sip
register 1000 auth-name "1000" password "1234"
At this point, we would likely need to see a "debug sip stack messages" to verify which device is sending the 403 message. If the Asterisk is sending the 403, you will likely need to check its log to determine why that message is sent. If the Adtran unit is sending the 403, typically the following debug output will allow us to determine the cause.
debug sip stack message
debug sip cldu
debug voice verbose
debug int fxs
debug isdn L2-formatted
Thanks for posting!
On Mon, Apr 9, 2012 at 2:47 PM, david
I went ahead and flagged this post as "Assumed Answered". If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons. This will make them visible and help other members of the community find solutions more easily. If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.