Extensions Unreachable???

Discussion in 'General' started by reynolwi, Jun 20, 2008.

  1. reynolwi

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    Im using elastix 1.0-17 and i have 3 grandstream gxp2000 phones. One phone works perfectly with no problems. I can call in and call out with no problems. The other 2 phones act like they arent even there. If i try and call the extension assigned to them i get the unavailable message, but yet i can call from them to the working phone and it rings. The phones show they are registered but i cant call them.

    Any ideas?
     
  2. rejil.rajan

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    Give the command sip show peer < extension of the phone that is not working >

    * Name : 200
    Secret : <Set>
    MD5Secret : <Not set>
    Context : from-internal
    Subscr.Cont. : <Not set>
    Language :
    AMA flags : Unknown
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 200@default
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 0
    Dynamic : Yes
    Callerid : "Name of the user" <200>
    Expire : 3432
    Insecure : no
    Nat : No
    ACL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Trust RPID : No
    Send RPID : Yes
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : IP Address of MachinePort 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 200
    SIP Options : (none)
    Codecs : 0x1c000c (ulaw|alaw|h261|h263|h263p)
    Codec Order : (ulaw,alaw,h263,h263p,h261)
    Status : Unmonitored
    Useragent : Grand Stream
    Reg. Contact : sip:200@<IP Address of the Phone>:5060;rinstance=98c2f50b705fde55

    If the IP address field is correct then this shows that the phone is registered. If you could sent the exact log when making a call to the device, I could tell you the exact issue
     
  3. reynolwi

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    Ok here is the information:

    This extension, 20301 (shown below) works perfect. You can call in, and call out to this extension.
    elastix*CLI> sip show peer 20301
    elastix*CLI>

    * Name : 20301
    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 : 20301@default
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <20301>
    MaxCallBR : 384 kbps
    Expire : 2534
    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 : 10.25.18.80 Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 20301
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : OK (45 ms)
    Useragent : Grandstream GXP2000 1.1.6.16
    Reg. Contact : sip:20301@10.25.18.80:5060;transport=udp


    These extensions, 30301 and 30302 (shown below) only let you call out from them. If you try and call the extension from a phone it rolls over to voicemail instantly like the phone isnt on.

    elastix*CLI> sip show peer 30301
    elastix*CLI>

    * Name : 30301
    Secret : <Not 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 : 30301@default
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <30301>
    MaxCallBR : 384 kbps
    Expire : 1767
    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 : 10.25.18.82 Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 30301
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : OK (25 ms)
    Useragent : Grandstream GXP2000 1.1.6.16
    Reg. Contact : sip:30301@10.25.18.82:5060;transport=udp




    elastix*CLI> sip show peer 30302
    elastix*CLI>

    * Name : 30302
    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 : 30302@default
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <30302>
    MaxCallBR : 384 kbps
    Expire : 1600
    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 : 10.25.18.81 Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 30302
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : OK (26 ms)
    Useragent : Grandstream GXP2000 1.1.6.16
    Reg. Contact : sip:30302@10.25.18.81:5060;transport=udp
     
  4. rejil.rajan

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    Hi

    Can you send me a log of the call that is being made from the working phone to the not working phone

    Please set the verbose as 3 and paste the log
     
  5. reynolwi

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    Ok i have no idea what happened, but i changed a different extension and now it works. I was messing with extension 50202 and changed the fax extension from freepbx to disabled and now for some reason it works. It still shows that the secret is not set for 30301 but it now says 30302 has the secret set.
     
  6. rejil.rajan

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    If you see the logs properly you might be able to see if the device is forwarding the calls to another extension, or if it is kept on DND. Please also ensure the same extensions are not part of any other application in Asterisk. Deleteing a recreating the extension should resolve the Not Set issue
     

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