VoIP Mailing List Archives

[ibc at aliax.net]

Hi, I've a SIP multi domain environment based on OpenSer, so subscribers use
OpenSer as registrar and proxy.

I want to use FreeSwitch (it's my first time with it) as an application
server / B2BUA, but I need it to work in SIP multi-domain way, this is, if
for example I use the voicemail server of FS, there will be voicemails for
users called:
- alice@domain1
- alice@domain2
- bob@domain1
...

This is unfeasible with Asterisk that just handles the username part of the
Request URI and From/To headers. How possible is it with FS? is it designed
to handle multidomain in SIP?

I have other questions more related to pure SIP protocol and FS as transparent
B2BUA:

alice ---(leg_a)--- FS (B2BUA) ---(leg_b)--- bob

- If alice adds a custom header ("Subject: hello!"), will this header appear
in leg_b to bob?

- If alice sends a custom in-dialog request (INFO request with no standar
Content-Body), will FS forward it to bob by just changing the needed data
(From_tag, To_tag, Call-ID, Via, CSeq...)?

- When alice sends the initial INVITE to bob (through FS), if bob replies
a "strange" response code ("497 Strange response"), will FS respect it and
forward upstream to alice?

- Is it possible FS to act as a SessionTimer endpoint with alice and bob after
the call (both legs) has been established? I don't want FS to handle media
(so I'll use "media_bypass" mode) but I need FS controling the call status by
sending in-dialog requests, hopefully using SIP Session Timers (RFC 4028).


Thanks a lot for all your replies, I hope to enjoy FS :)


--
Iñaki Baz Castillo

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