3000 can call 3001 but 3001 to 3000 is busy

Discussion in 'General' started by tording99z28, Jan 13, 2009.

  1. tording99z28

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    So I am new to Elastix and have been setting it up for the past couple days. I got 2 extensions setup on my local network with softphones on 2 seperate computers. Once softphone is setup as the ext 3000 and the other one is setup as 3001. I can sit at the 3000 softphone and call the 3001 and it works fine, rings, I can answer and talk and everything. However, if I call from the 3001 machine and try calling the 3000 machine I get a busy tone. Any idea why this is? Thanks in advance.
     
  2. soulofmischief87

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    hello when you set up the extensions did you make them both sip friends in type? another thing to check is if they have registered go to tools section and under cli try sip show peers and check if asterisk see's both of your extensions as being registered
     
  3. tording99z28

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    Yes they are both setup as friend type and when I do sip show peers I have both soft phones listed there and both their statuses are "OK".
     
  4. dicko

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    make sure there is either no firewall on the computer with softphone 3000 or if there is it is provisioned correctly. Swap the extensions (3000 -> 3001 and vice versa) check behavior again.
     
  5. tording99z28

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    Ok I did what you said and I swapped the 2. It is still the same, 3000 can call 3001 but 3001 can not call 3000 even though they are each on the other computer now. I would think this points towards the extension setup and not the computers themselves correct?

    I swapped them back to the original computers and still the same again, 3000 can call 3001 but 3001 gets busy signal when calling 3000.
     
  6. dicko

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    Indeed , delete both extensions and reprovision (use two completely different extensions just for shits and grins)
     
  7. soulofmischief87

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    also check what codec you are using if you are using a different codec, when ever ive used different codecs extensions cant call each other make sure you have maching codecs for both ends and i would also suggest you try a 3rd pc just to make sure u dont some kind of NIC problem
     
  8. tording99z28

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    Deleting the 2 extensions and re-adding them worked. I can now make internal calls from one machine to another and vice versa. Thanks guys.
     
  9. dicko

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    For completeness, next time use the Asterisk CLI and type sip show peer 3000, and sip show peer 3001, compare the two, any anomalies should be apparent, Asterisk will quite adequately transcode between any two codecs if the codec of the sender is not explicitly denied in the receiver (or vice versa) and the codec is "loaded" on the box (and the licensining if appropriate is valid)
     
  10. rafael

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    maybe you dial *78 by error and it activated do no disturb functionality

    regards,

    rafael
     

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