VoIP Blogging > Asterisk 1.6

Asterisk Telephony[Asterisk Telephony] It is worth noting as well the possibility of changing the timing for the SIP protocol, the so-called timers, defined in RFC 4028. This may require, for example, in cases where the network is characterized by stable long delay packages, as well as avoiding the hang SIP sessions, when communication with the SIP client fetches up during the session itself.

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VOIP-info.org Wiki Changeshttp://www.voip-info.org/wiki/view/RDNIS [VOIP-info.org Wiki Changes] RDNIS: by admin Fri 17 of Oct, 2008Depending on the number of lines you want to forward, you could use anything from a Linksys/Sipura 3000 for one line, Asterisk (or equivalent) for multiple lines, or as the previous reply suggested Quitum or other similar equipment could be used. Depending on the country you may be able to buy from existing service providers that already offer the service you described.

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