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my problem is this. I have google forward the call to gizmo5. I have this line in my sip file :<br><span>register => user:password@proxy</span>01.sipphone.com<br><div class="text">I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it?<br>At the end of my sip file i have this<br><br>[Calls-From-Gizmo-Network]<br>type=user<br>context=demo<br>disallow=all<br>allow=ulaw<br>allow=ilbc<br>allow=gsm<br>dtmfmode=rfc2833<br>host=proxy01.sipphone.com<br>insecure=very<br>username=user<br>secret=password<br>canreinvite=no<br><br>In my extentions i have this:<br>[fromgizmo]<br>exten => s,1,Wait(5)<br>exten => s,n,Answer<br>exten => s,n,Wait(2)<br>exten => s,n,Playback(welcome)<br>exten => s,n,Playback(test)<br>exten => s,n,Playback(test2)<br>exten => s,n,Hangup<br><br>The odd thing is i would have thought the <br>context=demo line from sip.conf would play the demo in extensions?<br>Instead it plays default which i put a line in to direct to fromgizmo...<br><br>[default]<br>;<br>; By default we include the demo. In a production system, you<br>; probably don't want to have the demo there.<br>;<br>include => fromgizmo<br>why not play demo?<br><br>Anyways, The first caller goes through just fine but the 2nd caller just gets a ringing. the output looks like this. <br><br><span> -- Executing [s@default:1] Wait("SIP/198.65.166.147-0</span><wbr><span class="word_break"></span>84fe8b0", "5") in new s tack<br><span> -- Executing [s@default:2] Answer("SIP/198.65.166.147</span><wbr><span class="word_break"></span>-084fe8b0", "") in new stack<br><span> -- Executing [s@default:3] Wait("SIP/198.65.166.147-0</span><wbr><span class="word_break"></span>84fe8b0", "2") in new s tack<br><span> -- Executing [s@default:4] Playback("SIP/198.65.166.1</span><wbr><span class="word_break"></span>47-084fe8b0", "welcome" ) in new stack<br><span> -- <SIP/198.65.166.147-084fe8</span><wbr><span class="word_break"></span>b0> Playing 'welcome' (language 'en')<br><span> -- Executing [s@default:5] Playback("SIP/198.65.166.1</span><wbr><span class="word_break"></span>47-084fe8b0", "test") i n new stack<br><span> -- <SIP/198.65.166.147-084fe8</span><wbr><span class="word_break"></span>b0> Playing 'test' (language 'en')<br><span> -- Executing [s@default:1] Wait("SIP/198.65.166.147-0</span><wbr><span class="word_break"></span>84fc2e0", "5") in new s tack<br><span> -- Executing [s@default:2] Answer("SIP/198.65.166.147</span><wbr><span class="word_break"></span>-084fc2e0", "") in new stack<br><span> -- Executing [s@default:3] Wait("SIP/198.65.166.147-0</span><wbr><span class="word_break"></span>84fc2e0", "2") in new s tack<br><span> -- Executing [s@default:4] Playback("SIP/198.65.166.1</span><wbr><span class="word_break"></span>47-084fc2e0", "welcome" ) in new stack<br><span> -- <SIP/198.65.166.147-084fc2</span><wbr><span class="word_break"></span>e0> Playing 'welcome' (language 'en')<br><span> -- Executing [s@default:5] Playback("SIP/198.65.166.1</span><wbr><span class="word_break"></span>47-084fc2e0", "test") i n new stack<br><span> -- <SIP/198.65.166.147-084fc2</span><wbr><span class="word_break"></span>e0> Playing 'test' (language 'en')<br><span> -- Executing [s@default:6] Playback("SIP/198.65.166.1</span><wbr><span class="word_break"></span>47-084fe8b0", "test2") in new stack<br><span> -- <SIP/198.65.166.147-084fe8</span><wbr><span class="word_break"></span>b0> Playing 'test2' (language 'en')<br><span> == Spawn extension (default, s, 5) exited non-zero on 'SIP/198.65.166.147-084fc2</span><wbr><span class="word_break"></span>e0'<br><span> -- Executing [s@default:7] Hangup("SIP/198.65.166.147</span><wbr><span class="word_break"></span>-084fe8b0", "") in new stack<br><span> == Spawn extension (default, s, 7) exited non-zero on 'SIP/198.65.166.147-084fe8</span><wbr><span class="word_break"></span>b0'<br><br>The
odd thing is that to asterisk it looks like both calls are taken right?
But whoever is the 2nd caller goes not get the call (it just rings and then goes to google voice mail). One more thing to
note is that if i make one call online (from sip softphone) and the other
from a land line or cell it works! Its only when i try to two "phones" (cell and/or land line) that it does not. How can i get two "phones" connected?<br>Thanks!</div><br /><hr />Windows Live™: Keep your life in sync. <a href='http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009' target='_new'>Check it out.</a></body>
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