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Thank you for your answer.<BR>
<BR>
Could you explain why the call fails ?<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Connected to Asterisk 1.4.25.1 currently running on asterisk (pid = 17936)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Verbosity is at least 25</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Core debug is at least 5</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- Executing [0473775006@intern:1] NoOp("SIP/twinkle-0a0567f8", "conversation to GSM") in new stack</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- Executing [0473775006@intern:2] Dial("SIP/twinkle-0a0567f8", "SIP/3starsnet/0473775006") in new stack</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- Called 3starsnet/0473775006</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- SIP/3starsnet-0a05c038 is making progress passing it to SIP/twinkle-0a0567f8</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">REGISTER attempt 1 to <A HREF="mailto:092779077@85.119.188.3">092779077@85.119.188.3</A></FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">REGISTER attempt 2 to <A HREF="mailto:092779077@85.119.188.3">092779077@85.119.188.3</A></FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Really destroying SIP dialog '1d9d81a23ac05e735663b37b750a640f@192.168.2.2' Method: OPTIONS</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- Got SIP response 500 "Service Unavailable" back from 85.119.188.3</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> -- SIP/3starsnet-0a05c038 is circuit-busy</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> == Everyone is busy/congested at this time (1:0/1/0)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> == Auto fallthrough, channel 'SIP/twinkle-0a0567f8' status is 'CONGESTION'</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Really destroying SIP dialog '340811e66bc43ba36fb5d507066fc1a7@192.168.2.2' Method: INVITE</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Really destroying SIP dialog 'xfdsxekzwoxcohv@localhost' Method: ACK</FONT></FONT><BR>
<BR>
<BR>
Jonas.<BR>
<BR>
<BR>
On Wed, 2009-06-24 at 02:47 +1000, Rob Hillis wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
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.
</PRE>
</BLOCKQUOTE>
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