On Sun, Jan 24, 2010 at 9:07 PM, Pizza Napoletana <pizzai...@gmx.com> wrote:
> This is my first time configuring sipx with an ITSP. I am sorry it is a 
> newbie question, but I don't see Speakeasy in the tested ITSP list. I'd like 
> to ensure that I configure and test it right (along the lines of a test case 
> spreadsheet I found on the sipx wiki) before I send my config and results to 
> the wiki.
>
> All I did was add a gateway and change just three items in the sipx GUI page 
> for SIP trunk ITSP Account page:
> 1. User name: <username>
> 2. Password: <secret>
> 3. User public address for call setup: Off  [** Our netscreen SSG's SIP ALG 
> is supposed to rewrite addresses, do firewall pinholes, etc.**]


I would recommend turning off the ALG. SipXbridge already does all of that.

>
> Did a quick test of outgoing and incoming calls. Seems to work fine.
>
> But Speakeasy gave me a whole bunch of parameters when they provisioned the 
> trunks. Here is what they gave (which I think is for asterisk):
>
> canreinvite=yes
> context=from-pstn
> dtmfmode=RFC2833
> insecure=very

No idea what "insecure=very" means but that does sound frightening.   :-)


> qualify=no
> host=<XXX>.speakeasy.net
> username=<XXX>
> secret=<XXX>
> type=peer
> Registration String: <username>:<secret>@<host>/<username>
>
> What I am supposed to do with all the other stuff Speakeasy gave me? Just 
> ignore it?



Try a few call flows. I suspect these are not relevant. If in and
outbound calling are working then chances are that you are OK.

>
> Thanks
>
>
>
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>
>
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>



-- 
M. Ranganathan
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