[Asterisk-Users] VoicePulse SIP

[Marc Storck]

asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP 
entries however....

so it is used to route via the Net.... if it cannot find a route via the 
Net or the link isn't working it will go to the next priority in your 
dialplan and do whatever you want, it doesn't re-configure your dialplan or 
route preferences.... let's say it's a bypass to IAX and SIP providers as 
it will tell you the username and server where users may be reached directly!!!

Marc

At 23:50 22.05.2004, you wrote:
>Andres wrote:
>
>>jparr at bgcfreedom.com wrote:
>>>Which providers give you a jitter buffer?
>>>
>>>
>>In Europe: VoipTalk and Magrathea.  In the US: Iconnecthere.   I am sure 
>>there are more.
>
>Clearpath gives jitter buffer as well.  http://www.clearpath1.com/
>
>John
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