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Thread: Asterisk PBX Integration Zimlet (new)

  1. #51
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    I went ahead and did the sip debug and these are what I believe would be the relevant lines for this extension. A lot of stuff scrolls so I just grabbed stuff that shows for that extension.

    Code:
    SIP Debugging enabled
      == Parsing '/etc/asterisk/manager.conf': Found
      == Manager 'administrator' logged on from 192.168.111.124
      == Manager 'administrator' logged off from 192.168.111.124
    
    <--- SIP read from 111.111.111.111 --->
    NOTIFY sip:our.server.com SIP/2.0
    Via: SIP/2.0/UDP 111.111.111.111:1025;branch=z9hG4bK-f63b27b4
    From: "My Name" <sip:100@our.server.com>;tag=c9023fb4dd76dbd1o0
    To: <sip:our.server.com>
    Call-ID: d29ded20-eaab2d55@192.168.10.100
    CSeq: 363888 NOTIFY
    Max-Forwards: 70
    Event: keep-alive
    User-Agent: Linksys/SPA962-5.2.8(SC)
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
    Sending to 70.131.148.35 : 1025 (NAT)
    
    <--- Transmitting (NAT) to 111.111.111.111:1025 --->
    SIP/2.0 489 Bad event
    Via: SIP/2.0/UDP 111.111.111.111:1025;branch=z9hG4bK-f63b27b4;received=111.111.111.111
    From: "My Name" <sip:100@our.server.com>;tag=c9023fb4dd76dbd1o0
    To: <sip:our.server.com>;tag=as241c3c1d
    Call-ID: d29ded20-eaab2d55@192.168.10.100
    CSeq: 363888 NOTIFY
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 0
    
    
    <------------>
    I don't see anything relating to the manager account except for the logon and logoff which seems strange.

  2. #52
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    I did not find any useful debugging command for the manager app yet. So far I used tcpdump But there is another thing to first. Please try to send an originate on asterisk cli. Example:
    Code:
    asterisk*CLI> originate sip/71 extension 100@intranet
    For your setting this could be something like:
    Code:
    asterisk*CLI> originate sip/100 extension 5555555@phones
    This is the equivalent to dialing using the Zimlet. If that also does not work we can isolate the problem to asterisk or phone related settings.

  3. #53
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    Default Sip Notify

    In your output it looks like your phone does not allow SIP NOTIFY. In the Zimlet i try to check the extension state first before placing the call. It seems that this is done using a SIP NOTIFY that your phone does not accept for some reason. So asterisk tells me that this extension does not exist. Maybe you 'll find a setting in your voip phone to accept SIP NOTIFY. Or you could try to test it with a Software SIP client first.

  4. #54
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    Using the originate command on the CLI the desk phone does get the call and when you pick up it shows it dialing out. However, the call doesn't reach the destination. For example, I set it to call my cell phone and it shows it as follows up to the point of me hanging up:

    Code:
    mail*CLI> originate sip/200 extension 5555555@phones
        -- Executing [5555555@phones:1] Set("SIP/200-006e1dd0", "CALLERID(ani)=1111111") in new stack
        -- Executing [5555555@phones:2] Dial("SIP/200-006e1dd0", "SIP/5555555@viatalk-2") in new stack
        -- Called 5555555@viatalk-2
      == Spawn extension (phones, 5555555, 2) exited non-zero on 'SIP/200-006e1dd0'
    That last line is me hanging up of course.

    On the other issue you mentioned, I tried using the Ekiga softphone with extension 200 and it still did not get the call just like the desk phone. I'll check into the sip notify feature on the phone.

  5. #55
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    hi bluecc
    Any news? I got it working with Ekiga Softphone. So i assume there are issues in your dialplan or asterisk config.

  6. #56
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    Hmm sadly there is at least one other person that as the same problem. There seems a problem with the extension check for certain phones or configurations. I don't now the exact problem yet. Maybe you could send me your sip.conf and extension.conf? As workaround try the older version 0.5. You should find it in the this Forum as attachment.

  7. #57
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    Default Hints!

    Finally got the problem! Actually for the ExtensionState action you need hints. So starting with Zimlet version 0.6 I added a feature to check the ExtensionState before placing the call. Seems that this is an old issue. Sorry!
    [Asterisk-Dev] ExtensionState problems using Manager.conf API
    So the easiest way ist to add
    Code:
    exten => 200,hint,SIP/200
    To your phones context.

  8. #58
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    I have tested your solution, in my dialplan i have the hint option, but still having the same trouble

  9. #59
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    Ok i created a Version 0.61 for you where you can disable the ExtensionState check.
    You need to add Property
    Code:
    <property name="astNoExtenCheck">true</property>
    to the Zimlets config.xml
    Please let me know if this works.
    Attached Files Attached Files

  10. #60
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    Default Get hints working

    In case you would still like to try the ExtenState stuff,
    you probably need to add some options to sip.conf for get hints working

    Code:
    [general]
    allowsubscribe = yes
    notifyringing = yes
    notifyhold = yes
    limitonpeers = yes
    You may check with cli core show hints
    Code:
    asterisk*CLI> core show hints
        -= Registered Asterisk Dial Plan Hints =-
                         82@hints               : SIP/82                State:Idle            Watchers  0
    ...
    You may also check Asterisk standard extensions - voip-info.org

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