Outbound call drops after 30 seconds

[vespino (Wesley van Kuijen)]

My FreePBX dropped it’s calls after 30 seconds, both inbound as outbound. I was able to fix the inbound calls by setting the correct local networks in SIP settings => NAT settings, but I’m struggling with the outbound calls. I have checked another FreePBX and applied the same settings, but without result.

NAT is set to “Yes”.

Am I looking in the right place?

[comtech]

This question pops up a lot on the forum, and usually is has something to do with router/NAT/firewall settings. I did a quick forum search and came up with these:

Calls dropping at 31 seconds

Vitelity is telling you that it’s changed its IP address. You’ll need to do a pjsip set logger on and then paste the traffic from vitelity to see what’s going on.

Fix Outbound calls drop after 30 seconds

I’ve run into this twice now on two different routers. The Sonicwall TZ170 and another Zyxel model. On an intermittent basis, outbound calls that route through these firewalls (and probably others) would simply drop after 30 seconds or so of successful two way audio. the issue turned out to be a default UDP timeout on the router. Changing the default from 30 seconds to 90 solved the problems. Interestingly, with one of the Sonicwall Tz170, after a power drop and reboot of the router the def…

Some outbound calls drop after 10-15 seconds

Is this X-Lite softphone ? If so, there is an "Options>advanced> Checkbox (in Times of network disruption, automatically hang up calls after X seconds… 30 seconds is the default. and the box is checked by default. This helped for me, the problem was on my softphone NOTE! Post edited to remove link to site that requires registration to show the answer. I also added the solution. mickecarlsson

There are a lot (a lot) more if you haven’t had the chance to review them.

[vespino (Wesley van Kuijen)]

I have found all of them and non of them seem to be related to my issue. Should I:

• Set RTP Timeout to a higher number?
• Set “Reinvite Behavior” to “yes” and (in my trunk setting) canreinvite=no => canreinvite=yes?
• Set NAT to “no”, to be honest: I don’t know what this does … NAT on the router is enabled?

[vespino (Wesley van Kuijen)]

I have set “RTP Timeout” and “RTP Hold Timeout” to 3600, now the calls seem to last longer than 30 seconds. I guess the RTP Timeout (which was set to 30) caused this issue. Hoping the outbound calls won’t be longer than 1 hour, this solved the problem I was having. If someone has a more durable solution, I’m open for suggestions. I have also mailed my provider for suggestions on this topic.

[vespino (Wesley van Kuijen)]

Apparently my solution did not work, so it’s back to the drawing board.

[Stewart1 (Stewart)]

If the audio is working (for the first 30 seconds), then the problem is unrelated to RTP.

Determine from the log whether the extension or the trunk is being dropped, and why. If it’s not clear, post a link to a paste of the log, including a SIP trace (mask credentials and other personal details as desired).

[vespino (Wesley van Kuijen)]

This is the log from around a call:

[edit]

[james (James Zhu)]

check ALG from your router and codecs as well.

[vespino (Wesley van Kuijen)]

Following an old post, I decided to reboot the server. You know what, it solved the issue. Think I’m going crazy.