[Asterisk-doc] Sip server

Brian Christie asterisk-doc@lists.digium.com
Thu, 27 May 2004 19:00:53 -0700


http://asterisk.org/index.php?menu=support

Juan G. Castaņeda wrote:
> Hi Greg,
> 
> Thanks for your email below. Could you please advise the email address 
> for the asterisk-user mailing list?. Thanks.
> 
> Regards,
> 
> Juan G. Castaņeda
> 
> 
> 
> Greg Varga wrote:
> 
>> Hi Juan,
>>
>> This is a mailing list to discuss topics about the Asterisk 
>> Documention project itself.  Unfortunately, we don't answer user 
>> related questions. :)
>>
>> Please redirect your question to the asterisk-user mailing list.
>>
>> Thanks,
>>   --Greg
>>
>> Juan G. Castaņeda wrote:
>>
>>>
>>> Sirs:
>>>
>>> 1.-  Can clients of a SIP server such as SER, use an Asterisk  SIP 
>>> gateway to give connectivity to the PSTN network?
>>>
>>> 2.-  Conversly, Calls that originate in the PSTN can traverse an 
>>> Asterisk SIP gateway to terminate at a SIP endpoint?
>>>
>>> 3.-  Finally,  may a call originate and terminate in the PSTN, but 
>>> cross a SIP Sever-based network in the middle?
>>>
>>> Could you please explain?
>>>
>>> Regards,
>>>
>>> Juan G. Castaņeda
>>>
>>> _______________________________________________
>>> Asterisk-Doc mailing list
>>> Asterisk-Doc@lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-doc
>>>
>>
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> 
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