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Old 31st Mar 2019, 2:32 am   #18
rambo1152's Avatar
Join Date: May 2010
Location: Manchester, UK.
Posts: 2,619
Default Re: Using a slide Switch 2A to switch between POTS and FreePBX

Originally Posted by julie_m View Post
While anybody's talking Asterisk, here's something I had to fix just this morning and seems appropriate to share.

If you get all calls, or all calls in one direction dropping after 32 seconds (billsec=32), check the external_media_address and external_signaling_address (yes, only one L) settings in your /etc/asterisk/pjsip.conf file match your external IP address.
On our systems that's in pjsip.transports_custom.conf and is managed by the GUI (Yes I know, GUIs are for WIMPs )
It's put my dyndns hostname in both positions.
Originally Posted by julie_m View Post
If you get some or most calls dropping after 32 seconds but some carrying on after that point, in both directions, then check your iptables firewall rules are not blocking any UDP ports in the RTP range used by your SIP trunk provider. (The range must end with an odd number, as far as Asterisk is concerned: it always uses two consecutive ports, for the two halves of the call. If you tell Asterisk 10000 - 20000, then it will actually use 10000 - 20001. If your provider doesn't use 20000+20001, set your Asterisk to use 10000 - 19999 instead.)

It's surprising just how many phone calls are actually connected for less than 32 seconds! Which is why it took so long to notice in the first place .....
Yes, I initially thought the system was working fine without forwarding the media ports until my wife informed me her calls were being cut off after 32 seconds (although she couched it in more threatening terms)
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