Thank you for your answer.
Could you explain why the call fails ?
Connected to Asterisk 1.4.25.1 currently running on asterisk (pid =
17936)
Verbosity is at least 25
Core debug is at least 5
-- Executing [0473775...@intern:1] NoOp("SIP/twinkle-0a0567f8",
"conversation to GSM") in new stack
-- Executing [0473775...@intern:2] Dial("SIP/twinkle-0a0567f8",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- SIP/3starsnet-0a05c038 is making progress passing it to
SIP/twinkle-0a0567f8
REGISTER attempt 1 to [email protected]
REGISTER attempt 2 to [email protected]
Really destroying SIP dialog
'[email protected]' Method: OPTIONS
-- Got SIP response 500 "Service Unavailable" back from 85.119.188.3
-- SIP/3starsnet-0a05c038 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/twinkle-0a0567f8' status is
'CONGESTION'
Really destroying SIP dialog
'[email protected]' Method: INVITE
Really destroying SIP dialog 'xfdsxekzwoxc...@localhost' Method: ACK
Jonas.
On Wed, 2009-06-24 at 02:47 +1000, Rob Hillis wrote:
> jonas kellens wrote:
> > Do you understand what is happening ?
> > I don't understand what this sentence means :
> > SIP/3starsnet-08d70ea8 is making progress passing it to
> > SIP/twinkle-08de0490
> Pretty simple really. Your SIP trunk 3starsnet is making progress with
> the call and Asterisk is passing that message on to SIP/twinkle.
> Entirely normal.
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