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
[email protected]: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 [email protected]: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
