[Asterisk-Users] VoicePulse SIP

Marc Storck mstorck at luxadmin.org
Sun May 23 00:19:42 MST 2004


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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