Hello list,
using Asterisk 1.4.30.
[Jun 16 21:35:12] -- Executing [s at sub-routing:12]
Dial("SIP/user110-0000005a", "SIP/user2|999") in new stack
[Jun 16 21:35:12] -- Called user2
[Jun 16 21:35:12] -- SIP/user2-0000005c is ringing
[Jun 16 21:36:12] WARNING[1991]: chan_sip.c:13073
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '0ae668e73053d17f33c852253f965683 at 192.168.1.150'. Giving
up.
[Jun 16 21:36:12] -- SIP/user2-0000005c is circuit-busy
[Jun 16 21:36:12] == Everyone is busy/congested at this time (1:0/1/0)
After exactly 60 seconds, the call is terminated, although I have given
a timeout-value of 999...
How come ??
Jonas.
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