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/1444514)
exten = 105,n,Hangup()
console output ...
-- Executing [5552530146@incoming:1] Answer(SIP/voipvoip.com-000f,
) in new stack
-- Executing [5552530146@incoming:2] Wait(SIP/voipvoip.com-000f,
1) in new stack
-- Executing [5552530146@incoming:3]
Playback(SIP/voipvoip.com-000f, beep) in new stack
-- SIP/voipvoip.com-000f Playing 'beep.alaw' (language 'en')
-- Executing [5552530146@incoming:4] Goto(SIP/voipvoip.com-000f,
105,105,1) in new stack
-- Goto (105,105,1)
-- Executing [105@105:1] Wait(SIP/voipvoip.com-000f, 2) in new
stack
-- Executing [105@105:2] Playback(SIP/voipvoip.com-000f,
hello-world) in new stack
-- SIP/voipvoip.com-000f Playing 'hello-world.alaw' (language 'en')
-- Executing [105@105:3] Dial(SIP/voipvoip.com-000f,
SIP/sip3.voipvoip.com/17037171624) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/sip3.voipvoip.com/1444514
[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-0010 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [105@105:4] Hangup(SIP/voipvoip.com-000f, ) in new
stack
== Spawn extension (105, 105, 4) exited non-zero on
'SIP/voipvoip.com-000f'
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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