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call cut after 30s


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32 seconds sounds a lot like a problem with the SIP routing, the ACK does not seem to make it. I would also think that it is a problem with the SIP router and/or firewall; maybe just a simple NAT problem: If a HTTP server works, it does not mean that the SIP server works as well...


I would make a PCAP trace for the call and take a look at it.

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Well the INVITE from OHV gets cancelled (BYE) after 478 ms. That does not leave much time for the PBX to answer. There is a Q.850 reason, something from the PSTN network.


What you can try to do is to back up the working directory of the PBX, upgrade to 5.5.0 demo license and see if the problem disappears.

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I would try to change the setup, for example try to run this from the home network instead of the office network. There are a lot of different router models out there, and lots of them are making trouble with SIP. For example, the FiOS router for Verizon has a "application layer gateway" built in that cannot be turned off and creates a lot of trouble. There is a good chance that you have a similar problem; the easiest way to find out is to change the setup and see where it makes a difference.

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