[Asterisk-Users] flashing zap using macro

[MJ]

I'm having problems using the following.

 

[sip]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
 
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup

 

I know I must be missing something simple, but here is the output from
dialing my home, answering the call, making another call to my home in order
to do a callwaiting transfer, doing a # and a *43016 (3016 is the sip number
I'm answering the Zap channel on).  I am using an analog phone off of a
Cisco ATA186 for extension 3016.  I replaced the number I was dialing from
into my home with 5555551212.

 

astera*CLI>

    -- Starting simple switch on 'Zap/1-1'

    -- Executing Wait("Zap/1-1", "1") in new stack

    -- Executing Answer("Zap/1-1", "") in new stack

    -- Executing DigitTimeout("Zap/1-1", "5") in new stack

    -- Set Digit Timeout to 5

    -- Executing ResponseTimeout("Zap/1-1", "10") in new stack

    -- Set Response Timeout to 10

    -- Executing Dial("Zap/1-1", "SIP/3014&SIP/3016&SIP/3017|35|tr") in new
stack

    -- Called 3014

    -- Called 3016

    -- Called 3017

    -- SIP/3017-2016 is ringing

    -- SIP/3016-4a65 is ringing

    -- SIP/3014-4269 is ringing

    -- SIP/3016-4a65 answered Zap/1-1

Jan 22 19:18:25 NOTICE[524306]: rtp.c:280 process_rfc3389: RFC3389 support
incomplete.  Turn off on client if possible

    -- Started music on hold, class 'default', on Zap/1-1

    -- Playing 'pbx-transfer' (language 'en')

    -- Stopped music on hold on Zap/1-1

    -- Executing Macro("Zap/1-1", "test|3016|5555551212") in new stack

    -- Executing Answer("Zap/1-1", "") in new stack

    -- Executing Flash("Zap/1-1", "") in new stack

    -- Flashed channel Zap/1-1

    -- Executing Dial("Zap/1-1", "SIP/5555551212|30|t") in new stack

Jan 22 19:18:48 WARNING[524306]: chan_sip.c:1114 create_addr: No such host:
5555551212

Jan 22 19:18:48 NOTICE[524306]: app_dial.c:673 dial_exec: Unable to create
channel of type 'SIP'

  == Everyone is busy at this time

    -- Executing Dial("Zap/1-1", "SIP/3016|30|t") in new stack

    -- Called 3016

    -- Got SIP response 486 "Busy Here" back from 192.168.0.232

    -- SIP/3016-f677 is busy

  == Everyone is busy at this time

    -- Executing Hangup("Zap/1-1", "") in new stack

  == Spawn extension (macro-test, s, 5) exited non-zero on 'Zap/1-1' in
macro 'test'

  == Spawn extension (sip, *43016, 1) exited non-zero on 'Zap/1-1'

    -- Hungup 'Zap/1-1'

    -- Starting simple switch on 'Zap/1-1'

Jan 22 19:18:58 NOTICE[540690]: chan_zap.c:4765 ss_thread: Got event 2
(Ring/Answered)...

    -- Executing Wait("Zap/1-1", "1") in new stack

    -- Executing Answer("Zap/1-1", "") in new stack

    -- Executing DigitTimeout("Zap/1-1", "5") in new stack

    -- Set Digit Timeout to 5

    -- Executing ResponseTimeout("Zap/1-1", "10") in new stack

    -- Set Response Timeout to 10

    -- Executing Dial("Zap/1-1", "SIP/3014&SIP/3016&SIP/3017|35|tr") in new
stack

    -- Called 3014

    -- Called 3016

    -- Called 3017

    -- SIP/3017-3574 is ringing

    -- SIP/3016-7c06 is ringing

    -- SIP/3014-9193 is ringing

    -- SIP/3016-7c06 answered Zap/1-1

  == Spawn extension (default, s, 5) exited non-zero on 'Zap/1-1'

    -- Hungup 'Zap/1-1'

astera*CLI>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050122/48f008b1/attachment.htm