<div dir="ltr"><div>This seems basic but something is missing.....</div><div> </div><div> </div><div>I dial from my cell phone to my DID and enter the context in extensions.conf</div><div>I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed.</div>
<div>But it fails. </div><div>And, I dpon't know why? Should I removed the Hangup application?</div><div>Syntax issue somewhere?</div><div> </div><div>I have a good SIP registration with the vendor, voipvoip.</div>
<div> </div><div>Thanks in advance for any feedback...</div><div> </div><div> </div><div> </div><div>[incoming]<br>exten => 5552530146,1,Answer()<br>exten => 5552530146,n,Wait(1)<br>exten => 5552530146,n,Playback(beep)<br>
exten => 5552530146,n,Goto(105,105,1)<br>;<br>;<br>[105]<br>exten => 105,1,Wait(2)<br>exten => 105,n,Playback(hello-world)<br>exten => 105,n,Dial(SIP/voipvoip/14445555514)<br>exten => 105,n,Hangup()<br></div>
<div> </div><div>console output .......</div><div> </div><div> -- Executing [5552530146@incoming:1] Answer("SIP/voipvoip.com-0000000f", "") in new stack<br> -- Executing [5552530146@incoming:2] Wait("SIP/voipvoip.com-0000000f", "1") in new stack<br>
-- Executing [5552530146@incoming:3] Playback("SIP/voipvoip.com-0000000f", "beep") in new stack<br> -- <SIP/voipvoip.com-0000000f> Playing 'beep.alaw' (language 'en')<br> -- Executing [5552530146@incoming:4] Goto("SIP/voipvoip.com-0000000f", "105,105,1") in new stack<br>
-- Goto (105,105,1)<br> -- Executing [105@105:1] Wait("SIP/voipvoip.com-0000000f", "2") in new stack<br> -- Executing [105@105:2] Playback("SIP/voipvoip.com-0000000f", "hello-world") in new stack<br>
-- <SIP/voipvoip.com-0000000f> Playing 'hello-world.alaw' (language 'en')<br> -- Executing [105@105:3] Dial("SIP/voipvoip.com-0000000f", "SIP/<a href="http://sip3.voipvoip.com/17037171624">sip3.voipvoip.com/17037171624</a>") in new stack<br>
== Using SIP RTP CoS mark 5<br> -- Called SIP/<a href="http://sip3.voipvoip.com/14445555514">sip3.voipvoip.com/14445555514</a><br>[Apr 9 16:07:11] WARNING[994]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission <a href="http://4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060">4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060</a> for seqno 102 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
Packet timed out after 32000ms with no response<br>[Apr 9 16:07:11] WARNING[994]: chan_sip.c:4198 retrans_pkt: Hanging up call <a href="http://4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060">4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060</a> - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
-- SIP/sip3.voipvoip.com-00000010 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br> -- Executing [105@105:4] Hangup("SIP/voipvoip.com-0000000f", "") in new stack<br> == Spawn extension (105, 105, 4) exited non-zero on 'SIP/voipvoip.com-0000000f'<br>
Asterisk*CLI><br></div></div>