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  #41 (permalink)  
Old 01-05-2009, 09:16 AM
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Posts: 72
Default Logs?

BTW: For better debugging also look in mailboxd.log. See BNC Business Network Communications AG - Asterisk PBX Integration Zimlet how to raise debug level for zimlet logs.
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  #42 (permalink)  
Old 01-05-2009, 09:49 AM
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Posts: 15
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This is the setting in sip.conf for extension 200

Code:
[200]
type=friend
nat=yes
canreinvite=no
insecure=very
host=dynamic
secret=****
username=200
context=phones
insecure=very
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  #43 (permalink)  
Old 01-05-2009, 01:16 PM
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Posts: 72
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hmm so phones seems to be definitively the right context setting. Could you please rise the zimlet debug level and post the releveant logs from mailbox.log?
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  #44 (permalink)  
Old 01-05-2009, 01:23 PM
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Posts: 15
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Sure thing. Can you refresh me on where to enable that and where mailbox.log would be? I'm relatively new to Zimbra so still getting the hang of things.

Thanks!
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  #45 (permalink)  
Old 01-05-2009, 01:27 PM
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Posts: 72
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You'll find the instructions at BNC Business Network Communications AG - Asterisk PBX Integration Zimlet section "Debugging", Subtitle "Serverside"
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  #46 (permalink)  
Old 01-05-2009, 01:39 PM
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Posts: 15
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Got that debug going and here's what I get in the log when I try to make a call. Note that I put 5555555 as the callee number here only.

Code:
2009-01-05 14:32:20,105 INFO  [btpool0-7] [] AuthProvider - Adding auth provider: zimbra com.zimbra.cs.service.ZimbraAuthProvider
2009-01-05 14:32:20,603 DEBUG [btpool0-7] [] zimlet - Getting ch_bnc_asterisk config
2009-01-05 14:32:20,603 DEBUG [btpool0-7] [] zimlet - {local={}, site={}, global={astManagerIp=our.server.com, enableSMS=true, astManagerPort=5038, astManagerUser=administrator, astSMSSendContext=sms-send, astSMSCenterChannel=CAPI/g1/0622100000, phonebookUrlCommonParams=ext=1, matchingRegExp=(\s|\t|^)\+?(\s?((\(\d{1,4}\))|(\d{1,4}))){6,10}(=?((\.|\,)?(\n|\r\n|\s|\t|$))), astDialChannelType=SIP, iddPrefix=00, astSMSVariable=SMS_MESSAGE, phonebookBaseUrl=http://tel.local.ch/q/, maxSMSLength=160, astDialContext=phones, astManagerSecret=OURPASS, phonebookUrlNumberParam=phone, srcPhonePrefix=, astActionTimeout=8000}}
2009-01-05 14:32:20,604 DEBUG [btpool0-7] [] zimlet - astManagerIp:our.server.com port:5038 user:administrator
2009-01-05 14:32:20,616 DEBUG [btpool0-7] [] zimlet - Encoding: UTF-8
2009-01-05 14:32:20,616 DEBUG [btpool0-7] [] zimlet - URL Params:
 info: null caller: 100 callee: 6306407999 smsFrom: null smsTo: null smsMsg:
 null
2009-01-05 14:32:20,616 INFO  [btpool0-7] [] zimlet - ch_bnc_asterisk caller: 100 callee 5555555
2009-01-05 14:32:20,616 DEBUG [btpool0-7] [] zimlet - Checking Extension state
2009-01-05 14:32:20,617 INFO  [btpool0-7] [] ManagerConnectionImpl - Connecting to our.server.com:5038
2009-01-05 14:32:20,661 INFO  [Asterisk-Java ManagerConnection-0-Reader-0] [] ManagerConnectionImpl - Connected via Asterisk Call Manager/1.0
2009-01-05 14:32:20,699 INFO  [btpool0-7] [] ManagerConnectionImpl - Successfully logged in
2009-01-05 14:32:20,779 INFO  [btpool0-7] [] ManagerConnectionImpl - Determined Asterisk version: Asterisk 1.4
2009-01-05 14:32:20,779 INFO  [btpool0-7] [] zimlet - Logged in asterisk manager.
2009-01-05 14:32:20,779 INFO  [btpool0-7] [] zimlet - Logged in asterisk manager.
2009-01-05 14:32:20,780 DEBUG [btpool0-7] [] zimlet - handleRequest ok
2009-01-05 14:32:20,781 DEBUG [btpool0-7] [] zimlet - {"callee":"5555555","extenState":-1,"extenStateWarning":true,"originateSuccess":false}
2009-01-05 14:32:20,789 INFO  [Asterisk-Java ManagerConnection-0-Reader-0] [] ManagerReaderImpl - Terminating reader thread: No more lines available: null
2009-01-05 14:32:20,790 INFO  [btpool0-7] [] ManagerConnectionImpl - Closing socket.
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  #47 (permalink)  
Old 01-05-2009, 01:49 PM
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Posts: 72
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So the number of your voip phone is still 200? Now it looks like its configured for number 100:
ch_bnc_asterisk caller: 100 callee 5555555
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  #48 (permalink)  
Old 01-05-2009, 02:03 PM
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Posts: 15
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That's correct. I'm at my office now and my exntension here is 100. I was using 200 earlier to test from my extension I use at home. However, both were not working. Hopefully that explains a little more. Sorry for not mentioning that.
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  #49 (permalink)  
Old 01-05-2009, 02:54 PM
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Posts: 72
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I can't find something wrong in zimlet logs So lets look more closely at asterisk to debug. Your voip phone is using SIP, right? So lets turn sip debug on. Use command
Code:
sip set debug
on the asterisk cli. Then try click-on-dial in zimbra again.
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  #50 (permalink)  
Old 01-05-2009, 03:26 PM
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Posts: 72
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Just to ensure there is not a handling problem. Normally the dialing should work that way:
  • Click phone number in Zimbra.
  • Wait until your phone rings. (Maybe message "Pickup phone to dial number #######" appears before ringing)
  • Pickup the phone to dial the clicked number
  • Message "Successfully dialed #######" appears
  • Call should take place
Mabybe you just picked up the phone before it rang?
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