Hope someone can provide a good learning resource for this, or pick up a few things I am doing wrong. I'm trying to setup FreePBX to test, but we are not replacing the current PBX. Disclaimer that my FreePBX may be all wrong, but judging by my debugging, the Adtran 908e is not right yet either.
We have a SIP trunk for local numbers and outgoing, and another SIP trunk for toll free.
I added voice trunk T04 for the new test PBX, changed the accept cost of a group of DID's, and created voice grouped-trunk SIP_GROUP_2 with the new trunk and lower accept cost. The trunks and grouped-trunks are as follows:
voice trunk T01 type sip
description "SIP TRUNK"
caller-id-override number-inbound 9 if-no-cpn
sip-server primary 192.168.1.10
sip-server secondary 192.168.1.20
check-supported replaces
!
voice trunk T02 type sip
description "Toll Free"
sip-server primary 67.x.x.x
authentication username "asdfg" password encrypted "asdfg"
!
voice trunk T03 type sip
description "Local DID"
sip-server primary 8.x.x.x
!
voice trunk T04 type sip
description "Test PBX"
sip-server primary 192.168.1.8
grammar from host local
transfer-mode network
!
voice grouped-trunk "SIP GROUP"
description "Production Servers"
trunk T01
accept $ cost 0
accept 555999640X cost 0
accept 555999641X cost 0
accept 555999642X cost 0
accept 555999643X cost 5
!
voice grouped-trunk LEVEL3_SIP
description "TOLL FREE"
trunk T02
accept 5554069600 cost 0
accept 18001234567 cost 0
accept $ cost 10
!
voice grouped-trunk SIP_LOCAL
trunk T03
accept $ cost 5
accept 411 cost 0
accept 911 cost 0
!
voice grouped-trunk SIP_GROUP_2
description "Test PBX"
trunk T04
accept 555999643X cost 0
I can list a lot more debugs, but this is the first Rx and Tx group on a call from FreePBX that fails:
192.168.1.21 is the Adtran IP.
208.x.x.x is our office IP
1234567890 is the cell phone I'm calling
1234561234 is our caller ID / main number
15:48:41.289 SIP.STACK MSG Rx: UDP src=192.168.1.8:5060 dst=192.168.1.21:5060
15:48:41.289 SIP.STACK MSG INVITE sip:1234567890@192.168.1.21:5060 SIP/2.0
15:48:41.289 SIP.STACK MSG Via: SIP/2.0/UDP 208.x.x.x:5060;rport;branch=z9hG4bKPj2bcc7676-3f47-46b8-87f4-ef4266811 957
15:48:41.290 SIP.STACK MSG From: <sip:1234561234@192.168.1.8>;tag=2f8cc17b-ea32-46d5-805b-0288b366ef16
15:48:41.290 SIP.STACK MSG To: <sip:1234567890@192.168.1.21>
15:48:41.290 SIP.STACK MSG Contact: <sip:asterisk@208.x.x.x:5060>
15:48:41.290 SIP.STACK MSG Call-ID: b360c72f-3a73-4d15-8599-e87fdf557d3a
15:48:41.290 SIP.STACK MSG CSeq: 26705 INVITE
15:48:41.290 SIP.STACK MSG Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGIS TER, REFER, MESSAGE
15:48:41.291 SIP.STACK MSG Supported: 100rel, timer, replaces, norefersub
15:48:41.291 SIP.STACK MSG Session-Expires: 1800
15:48:41.291 SIP.STACK MSG Min-SE: 90
15:48:41.291 SIP.STACK MSG Max-Forwards: 70
15:48:41.291 SIP.STACK MSG User-Agent: FPBX-13.0.192.9(13.14.0)
15:48:41.291 SIP.STACK MSG Content-Type: application/sdp
15:48:41.292 SIP.STACK MSG Content-Length: 312
15:48:41.292 SIP.STACK MSG
15:48:41.292 SIP.STACK MSG v=0
15:48:41.292 SIP.STACK MSG o=- 430554945 430554945 IN IP4 192.168.1.8
15:48:41.292 SIP.STACK MSG s=Asterisk
15:48:41.292 SIP.STACK MSG c=IN IP4 208.x.x.x
15:48:41.293 SIP.STACK MSG t=0 0
15:48:41.293 SIP.STACK MSG m=audio 10578 RTP/AVP 0 8 3 111 101
15:48:41.293 SIP.STACK MSG a=rtpmap:0 PCMU/8000
15:48:41.293 SIP.STACK MSG a=rtpmap:8 PCMA/8000
15:48:41.293 SIP.STACK MSG a=rtpmap:3 GSM/8000
15:48:41.293 SIP.STACK MSG a=rtpmap:111 G726-32/8000
15:48:41.294 SIP.STACK MSG a=rtpmap:101 telephone-event/8000
15:48:41.294 SIP.STACK MSG a=fmtp:101 0-16
15:48:41.294 SIP.STACK MSG a=ptime:20
15:48:41.294 SIP.STACK MSG a=maxptime:150
15:48:41.294 SIP.STACK MSG a=sendrecv
15:48:41.295 SIP.STACK MSG
15:48:41.299 SIP.STACK MSG Tx: UDP src=192.168.1.21:5060 dst=192.168.1.8:5060
15:48:41.300 SIP.STACK MSG SIP/2.0 100 Trying
15:48:41.300 SIP.STACK MSG From: <sip:1234561234@192.168.1.8>;tag=2f8cc17b-ea32-46d5-805b-0288b366ef16
15:48:41.300 SIP.STACK MSG To: <sip:1234567890@192.168.1.21>
15:48:41.300 SIP.STACK MSG Call-ID: b360c72f-3a73-4d15-8599-e87fdf557d3a
15:48:41.300 SIP.STACK MSG CSeq: 26705 INVITE
15:48:41.301 SIP.STACK MSG Via: SIP/2.0/UDP 208.x.x.x:5060;received=192.168.1.8;rport=5060;branch=z9hG4bKPj2bcc767 6-3f47-46b8-87f4-ef4266811957
15:48:41.301 SIP.STACK MSG Contact: <sip:1234567890@192.168.1.21:5060;transport=UDP>
15:48:41.301 SIP.STACK MSG Supported: 100rel,replaces
15:48:41.301 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
15:48:41.301 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.3.E
15:48:41.301 SIP.STACK MSG Content-Length: 0
I'm really confused why the Contact is our office IP with FreePBX, and the local IP for our production PBX. Maybe I have the firewall setup incorrectly on FreePBX? Besides that, do the trunks, groups, and accept lines look like they will work?
Here are the first few lines from a successful call from our production PBX:
15:54:57.235 SIP.STACK MSG Rx: UDP src=192.168.1.10:5060 dst=192.168.1.21:5060
15:54:57.235 SIP.STACK MSG INVITE sip:1234567890@192.168.1.21:5060 SIP/2.0
15:54:57.235 SIP.STACK MSG To: "City State" <sip:1234561234@192.168.1.21:5060>
15:54:57.236 SIP.STACK MSG From: <sip:1234561234@PBX.fqdn.com:5060>;tag=3hIbAzy
15:54:57.236 SIP.STACK MSG Via: SIP/2.0/UDP 10.100.124.10;branch=z9hG4bKlfnpvth2EbzfmAOytcgj
15:54:57.236 SIP.STACK MSG Call-ID: c41aaf0dc6279f489e53fbd738e77850@192.168.1.10
15:54:57.236 SIP.STACK MSG CSeq: 1 INVITE
15:54:57.236 SIP.STACK MSG Contact: <sip:1234561234@192.168.1.10>
15:54:57.236 SIP.STACK MSG Max-Forwards: 70