Posted: Thu May 27, 2004 9:14 pm Post subject: [Asterisk-doc] Sip server
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:
Quote:
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
>
Posted: Thu May 27, 2004 10:59 pm Post subject: [Asterisk-doc] Sip server
Quote:
-----Original Message-----
From: asterisk-doc-admin@lists.digium.com [mailto:asterisk-doc-
admin@lists.digium.com] On Behalf Of Juan G. Casta=F1eda
Sent: Thursday, May 27, 2004 5:15 PM
To: asterisk-doc@lists.digium.com
Subject: Re: [Asterisk-doc] Sip server
=20
Hi Greg,
=20
Thanks for your email below. Could you please advise the email address
for the asterisk-user mailing list?. Thanks.
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
>>
>
> _______________________________________________
> Asterisk-Doc mailing list
> Asterisk-Doc@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-doc
>
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