This is Asterisk configuration info and is usually added to the Asterisk 
sip.conf file - as Ranga said, as long as everything trunk related is 
working fine under sipX, you can safely disregard this information.

Thanks,
Jeff

On 25/01/2010 1:52 PM, M. Ranganathan wrote:
> On Sun, Jan 24, 2010 at 9:07 PM, Pizza Napoletana<[email protected]>  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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