[Asterisk-Users] Sip protocol question ...
Robert Rozman
rozman at fri.uni-lj.si
Fri Jan 7 15:47:03 MST 2005
----- Original Message -----
From: "Serge Schumacher" <serge at vonet.lu>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Sent: Friday, January 07, 2005 12:45 PM
Subject: RE: [Asterisk-Users] Sip protocol question ...
> What control is it ?
>
http://www.pernau.at/kd/voip/ActXPhone/
Regards,
Rob.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Robert
Rozman
> Sent: vendredi 7 janvier 2005 11:39
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sip protocol question ...
>
> Hi,
>
> I'm tryinig to debug SIP call from activex control based on MS RTC (A) to
> Asterisk (B). I use Etherreal to follow packages and I would like to ask
> short questions:
> - Session trace shows following order of packets:
> A - > B Invite
> B - > A 100 Trying
> B - > A 200 OK, with session description ; repeated 6
> times
> A - > B BYE sip: ....
> B - > A 200 OK
> - in my newbie logic it seems that B simply disconnects for some reason.
In
> session description there are codec specs. Unfortunately I don't have much
> docs on this active x control, so don't know how it behaves or whether it
> works.
>
> But anyway, does B anyhow tells reason why it requests disconnection ?
>
> Could I somehow from SIP packets gain knowledge about possible cause of
> disconnection ?
>
>
> Thanks in advance,
>
> regards,
>
> Robert.
>
>
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