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