My old config - T1 pri to Cisco T1 gateway - to freePBX (current updates applied per freePBX gui). I also have 4 channel SIPSTATION sip trunks. I have been running on this config for about a year.
Friday I did the cut-over from T1 pri to AT&T BVOIP trunks, including high speed Internet data circuit for our company. The cut-over went well, and works well - however, I for simplicity - and to verify AT&T was actually working - I temporarily disabled my Sipstation trunks during the cut-over. Now that I’m “on” the AT&T sip trunks, whenever I try to re-enable the Sipstation trunks, I get a number of errors - including the following:
Almost immediately I start losing Peers - ie. Notice  chan_sip.c: Peer ‘314’ is now lagged… followed by a message like Peer '313 now UNREACHABLE… the line buttons on my desk phone go RED, but after a few seconds they are back GREEN,
Logs show chan_sip.c --Registration for 'Le…(my Sipstation trunk).freepbx.org times out, trying again (attempt #2) and so on.
shortly later, start getting #1 above again… and my phone lines go RED again…
As soon as I disable my sipstation trunks I’m fine again… - phones don’t go offline, and no more errors in /var/log/asterisk/full.
This surprised me since the only thing I did to get the AT&T trunks added was to add them under TRUNKS and create a new outbound route. I would not expect enabling sipstation trunks to cause my handsets to go offline… ! I could understand that a conflict would make the sipstaion trunks unable to register or something - but enabling the sipstation trunks not only won’t register, but they seem to cause asterisk to start losing contact with my handsets!
Here are the settings for my AT&T trunk: (copied from someone’s how-to in the forum. TY!)
(there is one additional AT&T trunk configured the same but with a different IP.)
Additional notes: My freePBX server on a private network, with a pfSense router 1:1 NAT enabled for only AT&T and SIPSTATION IP addresses. To verify it wasn’t my router config, I moved my phone system outside the router - connected one NIC directly to my public IP address, (changed gateways, of course) but get the same results. AT&T sip trunks work great - but as soon as I enable my SIP trunks - same errors above #1,2,3. I did NOT try turning off the AT&T trunks and only run on sipstation trunks - because all my inbound calls now come over AT&T’s sip trunks
PS. I am running the freePBX distro, with Asterisk 11.21.2
Note sipstation is my backup SIP trunk so I am not actually down at the moment