How to connect Linksys PAP2 via Elastix

Discussion in 'General' started by eiger3970, Aug 24, 2009.

  1. eiger3970

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    Hi Elastix forumers...after many months I've got the gui...yippee...thanks to those who helped.

    Now I'm trying to access the outside world via my new elastix server. I currently run VoIP via my Linksys PAP2, but now want to run all calls via Elastix of course to play with all those cool new toys and gadgets available.

    I've got the X-Lite downloaded and run the extensions as per 'Elastix without tears'. When I dial 3001 from the X-Lite I get the message the extension cannot be reached.

    I think it's my cabling. I've got a cat 5 from my router to the pc running elastix. another cat5 from the router to the ata.

    I think this is what I need to work on next...cheers for any help.
    eiger3970.
     
  2. rafael

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    can you dial *43 and *65. With this info you would know if the phones are actually register.

    Regards,

    Rafael
     
  3. eiger3970

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    *43 works letting me hear my voice in the echo.
    *65 says my extension is 3001.
    When I dial extension 3001 it prompts for message bank. I leave a message but the message is not recorded as per Elastix dashboard 'Voicemail' which shows no voice mail. I can't even retrieve the voicemail.
    Dialling ext 3000, 3002, 3003, 3004, 3005, 3006 and 3007 says 'the person you are calling is unavailable, please try again' and shows 'Call failed: Service Unavailable' in X-Lite.

    Any quick tips on where to go in 'Elastix without tears' to solve this, and then move onto the 'real world'.
     
  4. eiger3970

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    OK, so I can operate internal calls to extensions 3001 and 3000. Although going to message bank doesn't work...but I'm more interested in connecting to the outside world.

    Have walked through all 'Elastix without Tear's and perhaps Chapter 7 ELASTIX ENDPOINT MANAGER is the problem?

    I run the endpoint scan and it finds no hardware. Is this right or wrong. I am using an ATA Linksys PAP2 so perhaps the endpoint only needs to find IP phones or pc cards as per Elaxtix without tears example/screenshot?

    I am running the internal 3000 and 3001 extensions via the ATA anyways, so I'm sure there's some connection via my Elastix server, then ATA and my phone.

    Any kind of scan to pick up what my be the error?
     
  5. danardf

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    This device should be never scanned by endpoint configurator.
    Remember that you must put 2 different Port sip on each line!
    for example: 5060 and 5061.

    you must be careful with the dial plan.
    In my case, I did put (xxxxxxxxxx.)
     
  6. eiger3970

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    Thank you danardf for the help...I am spending hours learning up all of this on Elastix without tears and the Elastix forum...I think I'm almost there now ;-)

    1.
    Panasonic phone base connected to
    Linksys PAP2 ATA Line 1.

    2.
    Linksys PAP2 ATA Line 1 and Line 2 registered to Elastix extension 201 and 202 respectively. (as per exact setup from Elastix without tears).

    3.
    Elastix SIP trunk and outbound route setup to VSP (as per exact setup from Elastix with tears).

    4.
    Call to outside world number gives the Elastix prompt
    'Please enter password followed by the pound key'.
    Password entered and hear Elastix prompt
    'Thank you, all circuits are busy now. Please try your call again later', and hangs up.

    5.
    a. I assume error is with Elastix trunk or outbound route setup
    , as I can dial the internal extensions fine.
    b. Cannot see if trunk or outbound route or extension are registered to VSP?
    c. Must be some report or error on Elastix to see if the dial plan or what problem could be happening?
    d. How does the exension 201 know to dial to the outside world via the trunk or outbound route? I cannot see the connection here as per Elastix without tears instructions?


    p.s.
    I tried putting the dial plan xxxxxxxxxx into the ATA Line 1. Same Elastix prompt
    'Please enter password followed by the pound key'.
    Password entered and hear Elastix prompt
    'Thank you, all circuits are busy now. Please try your call again later', and hangs up.
     
  7. eiger3970

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    OK, so I've played around with the sip_customer.conf file with different variations as per Elastix without tears. I think the call is trying to go to the outside world as asking for the password, is the password from the outbound route. The error after the correct outbound route password is entered on the phone is:
    'All circuits are busy now'.
     
  8. eiger3970

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    From bad to worse. Now the server is going wild giving crazy stats:
    CPU Info: GenuineIntel(R) Celeron(R) M processor 1.30GHz
    Uptime: 1 min
    CPU usage: 82.20% used of 1,296.81 MHz.
    Memory usage: 88.63% used of 216.79 Mb.
    Swap usage: 37.22% used of 447.99 Mb.

    Now I can't even log into my internal extensions on the Elastix server. (Extensions page won't display and just tries to load for ages).

    Uptime was a max of 8 days. Have reboot pc but still shows same crazy stats. PC is not that hot so who knows???
     
  9. danardf

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    Maybe a process loop?

    Make :
    # ps -ef | more

    And look at the 'TIME' result
     
  10. eiger3970

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    Tried the command, however I'm not getting many commands working in Elastix \Tools\Asterisk-Cli\Command and then Execute.
    Reinstalled Elastix and pc running ok again. Back to initial prob...internal extnsions all good, but outgoing to PSTN gives same error 'all circuits are busy now'. X-Lite will ring to VSP when configured. Tried multiple dial patterns.
    How can I run a 'debug' type command to find what the problem is?
     
  11. danardf

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    You can already give us your configuration!
    • Sip extension
      Trunk extension
      outgoing route

    Side PAP2, some config features on each line.

    Text format file attached please.

    If can use:
    CLI> sip debug (to start)
    CLI> sip no debug (to stop)

    Look at the result.

    For instant, I wait your config file.
     
  12. eiger3970

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    Thanks danardF. I ran the commands in my reinstalled Elastix 1.5.2-2

    as such but not sure where to find

    the information showing the possible

    errors?
    Elastix\PBX\Tools\Asterisk-

    Cli\Command
    sip set debug (SIP Debugging re-

    enabled).
    sip set debug off (SIP Debugging

    Disabled).

    Here's the configurations I have so

    far.

    I don't really have a SIP extension

    as such. I do have a trunk extension

    which is a SIP trunk.
    Dial Rules:
    XXXXXXXXXX
    PEER Details:
    username=094xxxxx
    type=peer
    secret=xxxxxxxx
    qualify=yes
    host=sip.gotalk.com
    fromuser=094xxxxx
    dtmfmode=auto
    disallow=all
    allow=ulaw (I have also tried

    allow=alaw).

    USER Details:
    username=094xxxxx
    type=user
    secret=xxxxxxxx
    qualify=yes
    insecure=very
    host=sip.gotalk.com
    fromuser=094xxxxx
    context=from-trunk
    canreinvite=no
    disallow=all
    allow=ulaw (I have also tried

    allow=alaw).

    Register String:
    094xxxxx:xxxxxxxx@sip.gotalk.com/094

    xxxxx


    Outbound Routes.
    Dial Patterns:
    XXXXXXXXXX

    Elastix\PBX\Tools|File

    Editor\File:sip_custom.conf
    useragent = whatever


    Linksys PAP2 ATA.
    Line 1 Dial Plan:
    *x|x.

    Preferred codec:
    G711a
     
  13. danardf

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    Try to use the same codec on any trunk and any extension!
    Into your dial plan from PAP2, you must try to put the same digits number!
    if you have the DP 1000, you have the digits like that: xxxx ou 1xxx.
    for external, if you must dial, 0225544661 ou must have: 0xxxxxxxxxx or 0.

    Must be careful to canereinvite = no and the port SIP must be different (PAP2)!
     
  14. eiger3970

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    2 problems now.
    1. Elastix is again overheating my pc/server. Memory usage: 92.47% used of 216.79 Mb. Can this be fixed by some sort of debug as I cannot reinstall Elastix everyday?

    2. Still outgoing calls getting error 'all circuits are busy'. I have changed sip.gotalk.com to the ip address 203.166.7.169 which seems to now allow the call to try and connect, but x-lite gives error 'call time out'.
     
  15. danardf

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    You have 256 Mo RAM, only?
    It's too short!

    I don't know if your trunk is good, but it's seems that a bad solution to use the 2 parts of trunk!

    In all my trunk I never use USER Details!

    You could try to remake your config trunk like this:

    PEER Details:
    • username=094xxxxx
      type=peer
      secret=xxxxxxxx
      qualify=yes
      host=sip.gotalk.com
      fromuser=094xxxxx
      dtmfmode=auto
      disallow=all
      allow=ulaw
      context=from-trunk
      canreinvite=no
      insecure=very

    094xxxxx : xxxxxxxx@sip.gotalk.com or
    094xxxxx : xxxxxxxx@sip.gotalk.com/094xxxxx

    Leave blank USER Details....
     
  16. itjumper

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    have you looked at the log file (/var/log/asterisk/full) ??
     
  17. danardf

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    That should be very hard if this person don't understand very well asterisk! ;)

    It's true, it's one solution! :huh:
     
  18. eiger3970

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    I have entered only user details in the trunk, and upgraded the 256Mb RAM to 512Mb RAM.
    I feel the Elastix software may be the problem as look at these crazy Elastix system stats even after the RAM upgrade.
    CPU Info: GenuineIntel Intel(R) Celeron(R) M processor 1.30GHz
    CPU usage: 99.32% used of 1,296.83 MHz
    Memory usage: 5.56% used of 470.60 Mb
    Swap usage: 10.16% used of 447.99 Mb
    Disk usage: 58%

    Now even the enternal extension won't register. Please help. Is there a command to run a debug check to find what the error could be?
     
  19. eiger3970

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    Crazy...now my pc is giving these stats with Elastix.
    Uptime: 1 day, 2 hour(s), 44 minute(s)
    CPU usage: 99.27% used of 1,296.83 MHz
    Memory usage: 93.20% used of 470.60 Mb
    Swap usage: 73.42% used of 447.99 Mb
    Hard drives: Capacity: 35.57GB
    Usage: 78%

    If I reinstall Elastix the software will run fine for a few days, but then same probs. What's going on???
     
  20. danardf

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    What did you install as module?
     

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