Mysterious "No matching peer found"

Discussion in 'General' started by bcnx, Aug 16, 2010.

  1. bcnx

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    Hi,

    I have these messages in /var/log/asterisk full after creating users:

    [Aug 17 11:21:36] NOTICE[5801] chan_sip.c: Registration from '"Test2" <sip:6001@192.168.1.15>' failed for '192.168.1.104' - No matching peer found



    SIP debug shows:

    <--- SIP read from 192.168.1.104:5062 --->
    REGISTER sip:192.168.1.15 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.104:5062;branch=z9hG4bK-6efe8437
    From: "Test2" <sip:6001@192.168.1.15>;tag=84254c8a41404373o2
    To: "Test2" <sip:6001@192.168.1.15>
    Call-ID: 50bde460-65999b4a@192.168.1.104
    CSeq: 16747 REGISTER
    Max-Forwards: 70
    Contact: "Test2" <sip:6001@192.168.1.104:5062>;expires=3600
    User-Agent: Linksys/SPA942-5.1.15(a)
    Content-Length: 0
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: replaces


    <------------->
    --- (12 headers 0 lines) ---
    Using latest REGISTER request as basis request
    Sending to 192.168.1.104 : 5062 (no NAT)
    elastix02*CLI>
    <--- Transmitting (no NAT) to 192.168.1.104:5062 --->
    SIP/2.0 404 Not found
    Via: SIP/2.0/UDP 192.168.1.104:5062;branch=z9hG4bK-6efe8437;received=192.168.1.104
    From: "Test2" <sip:6001@192.168.1.15>;tag=84254c8a41404373o2
    To: "Test2" <sip:6001@192.168.1.15>;tag=as10113507
    Call-ID: 50bde460-65999b4a@192.168.1.104
    CSeq: 16747 REGISTER
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0

    Makes sense, since "sip show peers" does not produce the newly added extension.
    The funny thing however is that "database show" in the Asterisk console is aware of this user:

    /AMPUSER/6001/cidname : Test2
    /AMPUSER/6001/cidnum : 6001
    /AMPUSER/6001/device : 6001
    /AMPUSER/6001/language :
    /AMPUSER/6001/noanswer :
    /AMPUSER/6001/outboundcid :
    /AMPUSER/6001/password :
    /AMPUSER/6001/recording : out=Adhoc|in=Adhoc
    /AMPUSER/6001/ringtimer : 0
    /AMPUSER/6001/voicemail : novm
    /DEVICE/6001/default_user : 6001
    /DEVICE/6001/dial : SIP/6001
    /DEVICE/6001/type : fixed
    /DEVICE/6001/user : 6001

    running /var/lib/asterisk/bin/retrieve_conf does not produce a particular error.

    I have to mention that this is an Elastix config running on two clustered nodes (Heartbeat and DRBD).


    Anyone any clue?

    thx!

    Bart
     
  2. bcnx

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    No hints, ideas? Anyone?
     
  3. bcnx

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    OK, did a third install on a new server and the same thing happens, so it does not seem to be clusterrelated. I'll check if there is a new version of Elastix or something.
     
  4. ramoncio

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    Hi bcnx,
    Which Elastix version do you have?
    Have you tried upgrading FreePBX from the unembedded FreePBX/Module admin page? You should try that.
    Could you explain a bit further your cluster setup?
    What do you have in your drbd partition?
     
  5. bcnx

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    Hi Raminoco,

    thank you kindly for your reply.
    It seems I did a major mistake while downloading and grabbed elastix 1.6 thinking it was Asterisk 1.6. I now came to my senses and am now installing Elastix 2.0. The cluster is up next, so I can report back in a little while with the new results.

    Watch this channel ;-)
     
  6. ramoncio

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    Hi,
    Elastix 1.6 is ok, and you might find more help and howtos with asterisk 1.4.
    In fact, most of us have our servers with Elastix 1.6, as Elastix 2.0 just got out, and the upgrade process is not trivial, to say the least.
     
  7. bcnx

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    Well, this is a fresh install. Would you advice against using 2.0 at this point?
     
  8. ramoncio

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    I haven't tested version 2.0 so much to use it on a production environment. And maybe I have to redo some of my post install script to work with Elastix 2.
    If it is a test server, you should try version 2, but in a production environment I would go for version 1.6, I would do some more testing into my environment before jumping in 2.0 version.
     
  9. bcnx

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    It is production, but for a rather small environment (6 phones). I suppose it is not Asterisk 1.6's stability you doubt, but the Elastix add-ons?

    Hmm, puts me kind of in a dilemma: would like to start learning about Asterisk 1.6 and solve my issue, but also would not like to set something up that has the potential to cause problems.

    B.
     
  10. bcnx

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    OK, quickly tried with 2.0 and I have the same problem, so that argument is down the drain.

    I've read some posts on the support towards Asterisk 1.4 and 1.6 and decided to stick with 1.4.

    Luckily I had reinstalled just one node and not both, so I need to redo just one.

    Will report back when I'm ready again to troubleshoot.
    I am somewhat suspecting my Linksys endpoints though I could not get a softphone registered either.

    What is Elastix's philosphy on storing users and extensions? It does not seem to use sip.conf and users.conf, but to either put them in the internal Asterisk database or mysql or both. Correct?
     
  11. ramoncio

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    Re:Mysterious

    Elastix uses FreePBX, it stores your extensions into the database, and it generates config files, usually with the _additional.conf suffix.
    You can also use unembedded freepbx, and maybe even upgrade it.
     
  12. bcnx

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    Thank you.

    Grepping on newly created users or extensions in /etc/asterisk does not match them however. They do show up while doing "database show" and with phpMyAdmin I could see them, so it seems that that part of the config is not put in conf files.

    I'm about to embark into another bigger Asterisk project (70 phones) and I was debating on using Elastix and Freepbx for that, but it though I heard it is possible to add your own programming, I guess it will be challenging to have it integrated with the freepbx stuff. Maybe a better idea to go the manual conf files-way for that ...

    cheers,

    B.
     

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