[asterisk-users] Dial with timeout don't end call

Magnus Benngård magnus.b at inputinterior.se
Sun Dec 13 07:25:39 CST 2009


Hi!

Trying to figure out what I am doing wrong...

1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.

extensions.conf
exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs)
exten => 975-INUSE,2,Hangup()
exten => 975-NOANSWER,1,VoiceMail(0317998975 at inputinterior.se,us)
exten => 975-NOANSWER,2,Hangup()
exten => 975-NOT_INUSE,1,Dial(SIP/0317998975&H323/00733025975 at Avaya,20)
exten => 975-NOT_INUSE,2,Goto(975-${DIALSTATUS},1)
exten => 975-NOT_INUSE,3,Hangup()

When calling 975, both SIP and cell phone starts to ring.
Answering on the SIP phone, cell phone stop ringing.
Answering on the cell phone, SIP phone keeps ringing.
If not answering any, cell phone stops ringing after 20 sec but
SIP phone just keeps ringing.

 == Using UDPTL CoS mark 5
 -- Executing [975 at inputinterior.se:1] Goto("SIP/0317998985-0000005e",
"975-NOT_INUSE,1") in new stack
 --
Goto (inputinterior.se,975-NOT_INUSE,1)
 -- Executing [975-NOT_INUSE at inputinterior.se:1]
Dial("SIP/0317998985-0000005e", "SIP/0317998975&H323/00733025975 at Avaya,20")
in new stack
 == Using UDPTL CoS mark 5
 -- Called 0317998975
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called 00733025975 at Avaya
 -- SIP/0317998975-0000005f is ringing
 -- H323/Avaya-16 is making progress passing it to SIP/0317998985-0000005e
 -- H323/Avaya-16 is making progress passing it to SIP/0317998985-0000005e
 -- H323/Avaya-16 is ringing
 -- Nobody picked up in 20000 ms
 -- Executing [975-NOT_INUSE at inputinterior.se:2]
Goto("SIP/0317998985-0000005e", "975-NOANSWER,1") in new stack
 -- Goto (inputinterior.se,975-NOANSWER,1)
 -- Executing [975-NOANSWER at inputinterior.se:1]
VoiceMail("SIP/0317998985-0000005e", "0317998975 at inputinterior.se,us") in
new stack
 --  Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
 --  Playing 'beep.gsm' (language 'se')
 -- Recording
the message
 -- x=0, open writing:
/var/spool/asterisk/voicemail/inputinterior.se/0317998975/tmp/EKTi4P
format: wav, 0x8c448d0
 -- User hung up
 == Spawn extension (inputinterior.se, 975-NOANSWER, 1) exited non-zero on
'SIP/0317998985-0000005e'

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