[Asterisk-Users] Asterisk as SIP <-> PSTN gateway

[Archie Cobbs]

Hi,

I'm new to Asterisk and have a couple of basic questions.

We're interested in using * simply as a SIP <-> PSTN gateway using
a T400P connected to one or more ISDN PRI lines (instead of using
a Cisco box which would cost more and come with no hackable source
code :-)

First, is Asterisk's SIP stack up to date and fully functional
with respect to the SIP protocol? Are there any known limitations
that would cause problems in this application? Is there a general
'to-do' list for SIP support?

E.g., it doesn't seem that chan_sip.c supports SUBSCRIBE/NOTIFY
DTMF events, which some Cisco boxes seem generate for DTMF (?).

Secondly, roughly what kind of CPU/system horsepower would required to
support transferring 96 channels of voice data between SIP/Ethernet and
the PRI if:

  (a) if no transcoding were being performed (i.e., both the
      RTP pkts and PRI B-channels were carrying ulaw data); and
  (b) transcoding from e.g. ulaw on the PRI <-> GSM in the RTP.

Thanks,
-Archie

__________________________________________________________________________
Archie Cobbs     *    Halloo Communications    *     http://www.halloo.com