8 thoughts on - Upgrade Asterisk 11 To 13 Or 11-16

• Hi,

  I did a talk on Astricon 2019 on this topic. Unfortunately there are no videos of that year but you can find my slides here covering some pitfalls. https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong

  Good luck by updating.

  BR
  Jöran

  schrieb am So., 6. Dez. 2020, 21:40:

• Is there more information how to implement it?

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• Hi,

  I guess describing how SIPp works here on a mailliste might be too much. But if you do not want to prove your setup automatically, you do not need to know SIPp.

  But there was a talk in 2014 Astricon giving an overview about SIP Testing with SIPp https://www.youtube.com/watch?v=TZMrPJM4HMc

  BR
  Jöran

• Thanks, yes initially it looked interesting. But I don’t see how “sipp”
  can be use to test my extension.conf dial plan.

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• Read UPGRADE.TXT in v13 and v16. Then read it again.

  I upgraded from Asterisk v11 to Asterisk v13. —
  http://help.nyigc.net/

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• Sound reasonable. I know it take time to debug the dial-plan after upgrade.

  Can I use sipp, from command line to call my local asterisk specific extension and to observe in another terminal via “asterisk -vvvvvvr”
  what it is doing?

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• I’m sure you can, but I’ve never done it.

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  http://help.nyigc.net/

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• Hi,

  Using SIPp to check your asterisk is working has some pitfalls. We recorded SIP invites (as these are the important parts of a call) of a normal call going through the Asterisk. We recorded it the old working way. In General we just compared what we had before with what we have after the upgrade. So if the Headers are Correct (from, to, our own X header, P
  Header etc). Since there are a couple of things you might want to check, e.g. someone is not allowed to place long distance calls, you can check the behavior of the asterisk as well so if the calls get rejected when some user dials some number.

  In SIPp there are these Scenario Files (XML Files) that contain a sequence of SIP Messages to send/receive. Using the receiving of Messages you can specifically check for presence or absence of a Header or a field in a header.

  there are lots of examples in the github repo https://github.com/SIPp/sipp

  For a A calls B call you need to start two SIPp Instances (one sending the call, one receiving the call)
  If your clients register to your Asterisk no not forget to do so, otherwise the Asterisk has no AOR to forward the call to. (Using plain UDP helps here a lot).

  The first check you build up might be some more work even if you never played around with SIPp but all what follows are quite simple and ensure quality.

  for my talk i put everything together you might need to place a simple call to an asterisk https://github.com/sipgate/signaling-test

  the start shell script will start first a registration to your Asterisk and then starts two sipp instances to place the call.

  feel free to use it.

  BR
  Jöran