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<font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
using Asterisk 1.4.30.<br>
<br>
[Jun 16 21:35:12] -- Executing [s@sub-routing:12]
Dial("SIP/user110-0000005a", "SIP/user2|999") in new stack<br>
[Jun 16 21:35:12] -- Called user2<br>
[Jun 16 21:35:12] -- SIP/user2-0000005c is ringing<br>
[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 '<a class="moz-txt-link-abbreviated" href="mailto:0ae668e73053d17f33c852253f965683@192.168.1.150">0ae668e73053d17f33c852253f965683@192.168.1.150</a>'. Giving up.<br>
[Jun 16 21:36:12] -- SIP/user2-0000005c is circuit-busy<br>
[Jun 16 21:36:12] == Everyone is busy/congested at this time (1:0/1/0)<br>
<br>
After exactly 60 seconds, the call is terminated, although I have given
a timeout-value of 999...<br>
<br>
How come ??<br>
<br>
<br>
Jonas.<br>
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