Sipstation trunks now break freepbx server after adding AT&T bvoip trunks

siptrunk
sipstation
configuration
Tags: #<Tag:0x00007fafc541c900> #<Tag:0x00007fafc541c6d0> #<Tag:0x00007fafc541c2c0>

#1

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:

  1. Almost immediately I start losing Peers - ie. Notice [2421] 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,

  2. Logs show chan_sip.c --Registration for 'Le…(my Sipstation trunk).freepbx.org times out, trying again (attempt #2) and so on.

  3. 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!)

videosupport=no
type=peer
srvlookup=no
sendrpid=no
realm=asterisk
qualify=2000
insecure=invite,port
host=12.xxx.xxx.xx9
dtmfmode=rfc2833
directrtpsetup=yes
context=from-pstn
canreinvite=yes
bindport=5060
allowoverlap=yes

(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


(Tony Lewis) #2

This is not a SIPStation issue. I bet your PBX can not resolve the DNS for SIPStation and Asterisk Chan_SIP when it can not resolve a FQDN locks up the SIP driver until it can resolve the FQDN.