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If my setup goes: Phone => asterisk =>
asterisk => PSTN termination provider
Can I define "canreinvite" on both asterisk boxes
so the phone call will go directly to the PSTN provider?
The Phone and first asterisk box are in my network,
then I am getting service from another guy running asterisk who is in turn
handing it off the the PSTN provider. He has "canreinvite" setup on his,
and I have it setup on mine. I see the RTP stream stop as soon as the call
is connected, but just wondering if his box is reinviting as well to go directly
from Phone to PSTN provider (since I can check his box to see if the RTP stream
is still going through it).
On a side note, with canreinvite enabled, sometimes
calls get dropped when the call is connected. Sometimes it doesn't do
this. Any ideas?
- Gabe
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