UK Vintage Radio Repair and Restoration Discussion Forum - View Single Post - Using a slide Switch 2A to switch between POTS and FreePBX

julie_m · 31st Mar 2019, 2:23 am

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.

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 .....

31st Mar 2019, 2:23 am	#17
julie_m Dekatron Join Date: May 2008 Location: Derby, UK. Posts: 7,735	Re: Using a slide Switch 2A to switch between POTS and FreePBX 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. 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 ..... __________________ If I have seen further than others, it is because I was standing on a pile of failed experiments.