Can't dial internal extensions (Elastix 1.5)

Discussion in 'General' started by rwbosveld, Mar 30, 2009.

  1. rwbosveld

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    Hi,

    I decided to try out Elastix 1.5 after setting up an Elastix 1.3 succesfully. I configured my voipbuster trunk. It is working perfectly, but I can't dial extensions. This is the output of Asterisk:


    dialparties.agi: Caller ID name is 'device' number is '100'
    dialparties.agi: USE_CONFIRMATION: 'FALSE'
    dialparties.agi: RINGGROUP_INDEX: ''
    dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 101 to extension map
    -- dialparties.agi: Extension 101 cf is disabled
    -- dialparties.agi: Extension 101 do not disturb is disabled
    > dialparties.agi: extnum 101 has: cw: 0; hascfb: 0 [] hascfu: 0 []
    > dialparties.agi: ExtensionState: 4
    dialparties.agi: Extension 101 has ExtensionState: 4
    -- dialparties.agi: Checking CW and CFB status for extension 101
    dialparties.agi: Failed to DbSet CALLTRACE/101 to 100 (0)
    -- dialparties.agi: Filtered ARG3: 101
    dialparties.agi: Setting default NOANSWER DIALSTATUS since no extensions available
    == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:4] NoOp("SIP/100-08a21650", "Returned from dialparties with no extensions to call and DIALSTATUS: NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/100-08a21650", "0?exit|return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/100-08a21650", "SV_DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/100-08a21650", "0?docfu|1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/100-08a21650", "0?docfb|1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/100-08a21650", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/100-08a21650", "Voicemail is novm") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/100-08a21650", "1?s-NOANSWER|1") in new stack
    -- Goto (macro-exten-vm,s-NOANSWER,1)
    -- Executing [s-NOANSWER@macro-exten-vm:1] NoOp("SIP/100-08a21650", "IVR_RETVM: IVR_CONTEXT: ") in new stack
    -- Executing [s-NOANSWER@macro-exten-vm:2] GotoIf("SIP/100-08a21650", "0?exit|1") in new stack
    -- Executing [s-NOANSWER@macro-exten-vm:3] PlayTones("SIP/100-08a21650", "congestion") in new stack
    -- Executing [s-NOANSWER@macro-exten-vm:4] Congestion("SIP/100-08a21650", "10") in new stack
    == Spawn extension (macro-exten-vm, s-NOANSWER, 4) exited non-zero on 'SIP/100-08a21650' in macro 'exten-vm'
    == Spawn extension (from-internal, 101, 1) exited non-zero on 'SIP/100-08a21650'
    -- Executing [h@from-internal:1] Macro("SIP/100-08a21650", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/100-08a21650", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/100-08a21650", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/100-08a21650", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/100-08a21650", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/100-08a21650", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/100-08a21650", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-08a21650' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-08a21650'

    The first lines indicate it can't set calltrace, but that shouldn't be a problem to ring that extension? What is this extensionstate (which is 4)?
     
  2. Patrick_elx

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    I had a problem kind of looking the same. All extensions were unreachable after using an elastix 1.4 file gui restore in 1.5.

    I solved it by going into unembedded freepbx and on each extension and ring group page individually click on submit. at the end, do a reload.


    tell me if it works for you
     
  3. Bob

    Bob

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    Rwbosveld,

    The extension states are as follows

    -1 = Extension not found
    0 = Idle
    1 = In Use
    2 = Busy
    4 = Unavailable
    8 = Ringing
    16 = On Hold

    Basically it is saying your extensions are unavailable...

    Can you confirm that your extensions have registered e.g. (under Asterisk CLI - sip show peers)

    If they show up, can you perform a sip show peer [EXTNUMBER] (replace extnumber with a valid ext) and post the info.

    Also have you confirmed that your codecs are in order. Were you using G729 on the 1.3 and forgotten to implement on 1.5?? This is typically the sort of thing that can cause this issue.

    Also did you try and restore anything, or did you hand code your trunks and extensions back in??

    Hope this helps a little

    Regards

    Bob
     
  4. rwbosveld

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    Hmm, I tried the unembedded freepbx, but it didn't work.

    Sip show peers tells me this:

    rwbosveld*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    Voipbuster/********** 194.120.0.198 N 5060 OK (42 ms)
    101 (Unspecified) D N 0 UNKNOWN
    100 (Unspecified) D N 0 UNKNOWN
    3 sip peers [Monitored: 1 online, 2 offline Unmonitored: 0 online, 0 offline]

    So the host is unspecified. Those extensions are registered, but somehow now available.

    sip how peer 100

    rwbosveld*CLI> sip show peer 100
    rwbosveld*CLI>

    * Name : 100
    Secret : <Set>
    MD5Secret : <Not set>
    Context : from-internal
    Subscr.Cont. : <Not set>
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 100@device
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <100>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : (Unspecified) Port 0
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username:
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :

    I think it has something to do with the IP adress I assigned to my Elastix box. I will try changing it. Codecs are properly configured. I did not restore a backup, it is done from scratch.


    [edit]

    Ok this is weird. Now it shows the peers are available, but it is still not working:

    rwbosveld*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    Voipbuster/********* 194.120.0.198 N 5060 OK (41 ms)
    101/101 192.168.1.254 D N 5060 OK (7 ms)
    100/100 192.168.1.254 D N 5060 OK (7 ms)
    3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline]


    Still extensionstate 4 :S

    Also registered an IAX extension with softphone Zoiper. It also has the same problem.

    Btw, if I drag an extension in FOP to another, the 'dragged' extension does ring. But then dialing to the other extension still fails. It must be something in Asterisk. :unsure:

    Thanks in advance. :)
     
  5. Bob

    Bob

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    rwbosveld,

    Your log is showing that Peer 100 is not registered.

    This needs to be resolved first. Confirm that the right IP address is being used in the phone (Soft or Hard phone), especially if you have changed the IP address of the Elastix Box.

    If you believe this is correct, as Patrick said, just go onto each Extension in Freepbx/Embedded Freepbx and press the SUBMIT button.

    Just one other thing, it may not make a difference, go onto the General page in Freepbx and allow anonymous SIP connections. Not something I normally advocate, but useful as a tool to try and resolve this issue.

    But basically, until you can see that the extension has registered, you will not have things working correctly.... (E.g. STATUS needs to not show UNKNOWN)

    Regards

    Bob
     
  6. perezil

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    OK, I am experiencing anomalies also. I was on 1.5-6 and tried to backup the data on it and then restore it on 1.5-9. That didn't go well at all, so I decided to install 1.5-9 from scratch.

    I too am having the Outbound Calls are OK, but Inbound calls are lost. Everything goes to voice mail and none of the lines are dialable neither internally nor externally.

    I even installed the XTEN client on my notebook. It can dial out, but dialing it from another extension I get the "unavailable" message.

    W-e-i-r-d!


    I suspect a strange little bug in here.

    (being new to the Asterix/Trixbox/FreePBX/Elastix, I am spinning my wheels and getting tiered)

    FYI: My VoIP provider is Broadvox and I authenticate by IP Address.

    Luis
     
  7. rwbosveld

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    I think also it's a bug. My extensions are registered bob :)
    I found the iso with elastix 1.3, I installed it and now my extensions are working fine. I think this must prove it's a bug.

    edit:

    Ok, with 1.3 I can't dial out with voipbuster, but the config is the same. :woohoo:
     
  8. perezil

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    The last build that worked for me was 1.5-6
     
  9. rwbosveld

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    Already fixed the dialout problem. Asterisk tried to force the use of GSM codec. After putting disallow=gsm in the context of the voipbustertrunk that is fixed.
     
  10. ermali

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    you have to fill the CID NUM ALIAS ...i have inserted the same number like extension and it worked.
     
  11. robclay

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  12. rafael

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    Can you please type again the url. It seems to be a problem with it
     
  13. robclay

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    No problem,

    See the Similar Thread showing how the problem was fixed for me.
     
  14. perezil

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    This sounds like a very possible fix (one I didn't try). I thought I had set up the disallow=all, followed by allow=ulaw. Maybe I missed it.

    I'll give it a try in a few days. I am currently BETA testing Thirdlane PBX, so I won't be able to get back to it until this weekend.
     
  15. perezil

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    ermali, you are partially correct. However, follow rwbosveld's post FIRST. THEN try your CID NUM ALIAS test. Depending how your provider, you may or may not need it. It's one of this annoyances...
     

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