The Adtran community holiday season is starting next week! The holiday period will span from December 21, 2024 to January 6, 2025. During this time, responses to feedback form submissions may be delayed. If you are encountering product issues, you can reach out to Adtran support at any time.
cancel
Showing results for 
Show  only  | Search instead for 
Did you mean: 
sroberts
New Contributor

Adding a 2nd PBX

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

Labels (1)
0 Kudos