On 09.04.2013, at 23:12, Thomas Perron <thomas.per...@gmail.com> wrote:
> 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>
Enter "sip set debug on" at the console and show us the output from the call
attempt (you should get a log of your SIP traffic together with the "normal"
console output).
--
marie
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