Asterisk dropping STD calls


I’ve had to pick up the maintenance of an asterisk system in the past few weeks due to the original person who installed it leaving the company. I’m pretty green at this asterisk stuff, so I’m sorry in advance if I make any newbie mistakes.

My problem is that when a phonecall comes in to Site A and then gets tranferred over the VPN to Site B, after about 8-10 minutes the call gets dropped.

As i mentioned above, I’m pretty new at this and don’t really know where to start looking. If you need any conf files or questions answered, I’ll do my best to answer them.

Also as an aside and not really important…is it possible to set a different ring tone for incoming line phone calls as opposed to internal transferring between sites?

Thanks in advance.

What protocol is being used?

As a first step, set up an interactive console with Asterisk -r and issue “core set verbose 5” (or higher). Then report what, if anything, gets logged at the point the call fails. You may also need to turn on protocol specific tracing (or use a network trace tool, e.g. wireshark).

The answer for ring tones depends on the protocol used, and can also depend on the type of phone. For SIP phones, you need to add a custom SIP header, that is phone specific.

The protocol for the VPN? That is OpenVPN running on UDP. Anything else as far as I’m aware is TCP.

I’ve spoke with the site that has the troubles in further detail and even though they’re not in until next Wednesday, it seems to happen after 8-10 minutes.

So next Wednesday I shall post the CLI logs you requested.

For the VPN: SIP, IAX, or SCCP, and as the term VPN, for phone networks, predated IP telephony, it could also be Q.SIG, SS7, DPNSS, etc.

That for the public side may also be relevant, which almost certainly would exclude SCCP and would probably exclude IAX, but could include various analogue line signalling systems.