Calls dropping at around 15 minutes

Hi There

I am running FreePBX Distro 14 on Asterisk 11

We are having dropped calls at around the 15 minute mark. It isn’t at 15 minutes exactly, usually around 15:20 or something like that. It appears to be both inbound and outbound.

In Asterisk SIP Settings under Chan SIP Settings, I have session-timers=refuse entered into other SIP settings, as well as session-timers=refuse entered into my trunk context and the 15 min call drops continue to happen.

I escalated to my carrier, who ran a trace and they are saying RTP stopped transmitting from our switch, so they don’t think it would be a refresh timer on their side – not to mention this issue persists when using multiple carriers from our testing.

What else should I be looking for?

Thank you!

Search the forums for “calls dropping 30 minutes” and you’ll see lots of problem reports that sound just like yours. The solutions you’ll find there (adjusting the timers in your firewall, routers, and the “qualify=” settings) should resolve your issue.

Advanced Settings>Dialplan and Operational>Asterisk Dial Options… should be set to"Ttr" Here you can set up call duration timer, if you know what you are doing

Is the carrier FlowRoute?

I am still having this issue, and any suggestions would be greatly appreciated as I’m not sure what to check at this point.

As I stated in my original post, FreePBX appears to be dropping Outbound calls only at around the 15:10 mark. I have set session-times=refuse under CHAN SIP SETTINGS as well as session-timers=refuse entered into my trunk context and the issue still persists.

My RTP times under CHAN SIP SETTINGS are set to the following…

RTP Timeout: 30
RTP Hold Timeout: 3600
RTP Keep Alive: 60

Also as previously stated the issue happens across multiple carriers so appears to be switch related.

Any input would be greatly appreciated as I’m totally stumped at this point. I have multiple other FreePBX switches in production that have never had this problem before.

Thank you!

Any luck on this issue Kwriley87?

What devices are being used on both ends? I had a similar issue and after lots of frustration and testing, I determined it was a remote ATA device that had a silence setting that was too low.

@kwriley87 I had a similar issue—-we had a hell of a time catching Malformed SDP on reinvites which was causing our carrier to send 400 and FPBX to hangup.

Don’t know what’s causing it here but if you’re still at a loss take a look at SDP body specifically the o and c fields if your carrier is utilizing or supporting session timers.