asterisk users - Apr 2013 - Connect to an outbound channel and dial a phone number??

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 at incoming:1]
Answer("SIP/voipvoip.com-0000000f", "") in new stack
    -- Executing [5552530146 at incoming:2]
Wait("SIP/voipvoip.com-0000000f",
"1") in new stack
    -- Executing [5552530146 at incoming:3]
Playback("SIP/voipvoip.com-0000000f", "beep") in new stack
    -- <SIP/voipvoip.com-0000000f> Playing 'beep.alaw' (language
'en')
    -- Executing [5552530146 at incoming:4]
Goto("SIP/voipvoip.com-0000000f",
"105,105,1") in new stack
    -- Goto (105,105,1)
    -- Executing [105 at 105:1] Wait("SIP/voipvoip.com-0000000f",
"2") in new
stack
    -- Executing [105 at 105:2] Playback("SIP/voipvoip.com-0000000f",
"hello-world") in new stack
    -- <SIP/voipvoip.com-0000000f> Playing 'hello-world.alaw'
(language
'en')
    -- Executing [105 at 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 at 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 at 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 at 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>
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On 09.04.2013, at 23:12, Thomas Perron <thomas.perron at 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 at incoming:1]
Answer("SIP/voipvoip.com-0000000f", "") in new stack
>     -- Executing [5552530146 at incoming:2]
Wait("SIP/voipvoip.com-0000000f", "1") in new stack
>     -- Executing [5552530146 at incoming:3]
Playback("SIP/voipvoip.com-0000000f", "beep") in new stack
>     -- <SIP/voipvoip.com-0000000f> Playing 'beep.alaw'
(language 'en')
>     -- Executing [5552530146 at incoming:4]
Goto("SIP/voipvoip.com-0000000f", "105,105,1") in new stack
>     -- Goto (105,105,1)
>     -- Executing [105 at 105:1] Wait("SIP/voipvoip.com-0000000f",
"2") in new stack
>     -- Executing [105 at 105:2]
Playback("SIP/voipvoip.com-0000000f", "hello-world") in new
stack
>     -- <SIP/voipvoip.com-0000000f> Playing 'hello-world.alaw'
(language 'en')
>     -- Executing [105 at 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 at
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 at 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 at 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