Well, it was my assumption that the call was ultimately routed through
the PSTN -- that's the root of my curiosity.
I have access to the remote PBX, so I threw tcpdump on both ends and
made a call... Very interesting stuff for sure!
*Egress (my PBX):
*
From: "sipxbridge" <sip:####[email protected]>;tag=...
*Ingress (their PBX):
*
From: "sipxbridge" <sip:####[email protected]>;tag=...
Their side uses Asterisk and their upstream trunk provider is using
Asterisk, so there's not much I can see otherwise on their end. However,
on my end -- I can see the media stream is going to O1. What's
interesting about that, is that the DID I was calling is serviced by O1
for origination, and that particular call path goes to my provider who
resells O1.
So it looks like it was SIP all the way, never touching the PSTN.
Interesting how well it worked. I need to enable g722 on my Aastra and
see if it and their Polycom 550 will do HD Voice (I know g722 is enabled
on the Polycoms).
-- Robert
On 1/8/2010 8:25 PM, Tony Graziano wrote:
Its not that abvious.
It depends on the identity sent through your ITSP. I'm assuming in
this case PSTN means a sip trunk, because most analog providers do not
allow you to assert the caller-d (some do on PRI or ISDN, but not
POTS). Since yours says sipxbridge, it must be something to do with
the caller-id being sent through sipXbridge. I would look at the
p-asserted-identity.
Use default asserted identity (Default: checked)
If checked (default), use the default asserted identity. Otherwise,
you must enter a usern...@domain to override the default.
Asserted identity
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