Did move 0317998975 phone from my home to my office, didnt need any:
nat=yes in sip.conf, everything worked.
I did also add callcounter=yes in sip.conf so I am not sure how it
will work when I move the phone to my home and need nat=yes again.
Will do some tests later tonight when I am at home.
On Sun, 13 Dec 2009 14:25:39 +0100, Magnus Benngård wrote: 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-${DEVICE_STATE(SIP/0317998975)},1)
exten => 975-INUSE,1,VoiceMail([email protected],bs)
exten => 975-INUSE,2,Hangup()
exten => 975-NOANSWER,1,VoiceMail([email protected],us)
exten => 975-NOANSWER,2,Hangup()
exten => 975-NOT_INUSE,1,Dial(SIP/0317998975&H323/00733025...@avaya,20)
exten => 975-NOT_INUSE,2,Goto(975-${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 [[email protected]:1] Goto("SIP/0317998985-0000005e",
"975-NOT_INUSE,1") in new stack
-- Goto (inputinterior.se,975-NOT_INUSE,1)
-- Executing [[email protected]:1]
Dial("SIP/0317998985-0000005e", "SIP/0317998975&H323/00733025...@avaya,20")
in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- Requested transfer capability: 0x00 - SPEECH
-- Called 00733025...@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 [[email protected]:2]
Goto("SIP/0317998985-0000005e",
"975-NOANSWER,1") in new stack
-- Goto (inputinterior.se,975-NOANSWER,1)
-- Executing [[email protected]:1]
VoiceMail("SIP/0317998985-0000005e", "[email protected],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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