[Freeswitch-users] Can I get SIP DID working?

[Brian West]

You'll need to setup an ACL to let them in without authentication.   
Look at the Asterisk to FreeSWITCH section on the wiki.

http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

/b

On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote:

> I have used Freeswitch with a DID from Voxbone working on IAX, and
> that have been working so well. But now Voxbone will discontinue the
> IAX service, so I have to get DID working on SIP, and I am wondering
> if that will work at all in my configuration.
>
> Freeswitch is behind nat, so when I get a call from Voxbone, see http://pastebin.freeswitch.org/4629
>  ,
> it will come in on port 5060. Have you covered this situation in the
> documentation?
>
> Ivan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Brian West
sip:brian at freeswitch.org