Re: [asterisk-users] Reinvites and SIP/RTP

2008-07-16 Thread Martin Schuhmacher 
Noah Miller schrieb:
 Hi Adrian -
 
 When I use re-invite, does the Asterisk server stay in the SIP conversation,
 and just RTP traffic diverts, or does the SIP transfer away from the A*k
 server too ?
 
 I'm sure somebody will correct me if this is wrong, but I believe the
 signalling must stay with asterisk, as asterisk needs to know if it
 should provide any services for the call (music on hold, transfer,
 etc).

yes, 'only' rtp goes direct, SIP stay on asterisk since it might
be a hangup or something else comes in.

Yours
Martin

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[asterisk-users] Reinvites and SIP/RTP

2008-07-15 Thread Adrian Marsh 
Hi All,

 

When I use re-invite, does the Asterisk server stay in the SIP
conversation, and just RTP traffic diverts, or does the SIP transfer
away from the A*k server too ?

 

Thanks,

 

Adrian

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Re: [asterisk-users] Reinvites and SIP/RTP

2008-07-15 Thread Noah Miller 
Hi Adrian -

 When I use re-invite, does the Asterisk server stay in the SIP conversation,
 and just RTP traffic diverts, or does the SIP transfer away from the A*k
 server too ?

I'm sure somebody will correct me if this is wrong, but I believe the
signalling must stay with asterisk, as asterisk needs to know if it
should provide any services for the call (music on hold, transfer,
etc).


- Noah

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