FreePBX Anveo configuraiton on Static IP

douglasmv · December 15, 2025, 3:37am

I fixed the issue with on the Anveo side but still having issues on the freepbx side. I did set the Ip address to match and its still not working it rings once and thats it. Im going to check my anveo logs. For some reson they are not showing to give me live stats I have to wait.

douglasmv · December 15, 2025, 3:40am

What needs to be in the cliant URI? and Server URI?

Stewart1 · December 15, 2025, 3:54am

Nothing. Those settings are related to registration, which you are not using.

Stewart1 · December 15, 2025, 3:57am

Since the call is making it to Asterisk, turn on pjsip logger, make a failing test call in, paste the Asterisk log for the call at pastebin.com and post the link here.

douglasmv · December 15, 2025, 4:04am

/*>>>|174.141.213.62:5060 @ 2025-12-15 03:47:56 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK592a6d44db115d52c1c798de94db5b26;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c3344fca6d07cf9be79fde1bed786c0f
To: <sip:[email protected]>
Call-ID: [email protected]_1-b2b_1
CSeq: 200 INVITE
Contact: Anonymous <sip:[email protected]:5060>
Expires: 300
User-Agent: ACC
cisco-GUID: 27669653-298810312-190787488-2356164701
h323-conf-id: 27669653-298810312-190787488-2356164701
P-Asserted-Identity: <sip:+13363744673;[email protected]:5060>
P-attestation-indicator: A
P-origination-id: uuid:561517dd-c7a2-4380-84df-036b0334dec6
X-anveo-e164: 13364960050
Content-Type: application/sdp
Content-Length: 281

v=0
o=Sonus_UAC 598746 621471 IN IP4 67.231.1.112
s=SIP Media Capabilities
c=IN IP4 67.231.1.83
t=0 0
m=audio 50614 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

/*<<<|174.141.213.62:5060 @ 2025-12-15 03:47:56 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK592a6d44db115d52c1c798de94db5b26
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=c3344fca6d07cf9be79fde1bed786c0f
To: <sip:[email protected]>
CSeq: 200 INVITE
Server: FPBX-17.0.23(22.6.0)
Content-Length:  0


/*<<<|174.141.213.62:5060 @ 2025-12-15 03:47:56 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK592a6d44db115d52c1c798de94db5b26
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=c3344fca6d07cf9be79fde1bed786c0f
To: <sip:[email protected]>;tag=57e83ab9-6759-4d0f-a045-71829ebfc85c
CSeq: 200 INVITE
Server: FPBX-17.0.23(22.6.0)
Contact: <sip:174.141.213.62:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   280

v=0
o=- 598746 621473 IN IP4 174.141.213.62
s=Asterisk
c=IN IP4 174.141.213.62
t=0 0
m=audio 11966 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

/*>>>|174.141.213.62:5060 @ 2025-12-15 03:47:56 */
ACK sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;rport;branch=z9hG4bK2c1aa7deab5a9ffd7a862129545fd387
Max-Forwards: 70
From: <sip:[email protected]>;tag=c3344fca6d07cf9be79fde1bed786c0f
To: <sip:[email protected]>;tag=57e83ab9-6759-4d0f-a045-71829ebfc85c
Call-ID: [email protected]_1-b2b_1
CSeq: 200 ACK
User-Agent: ACC
Content-Length: 0


/*>>>|174.141.213.62:5060 @ 2025-12-15 03:48:02 */
BYE sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK760f485af4f5b1610e169a69cbe4c62c;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c3344fca6d07cf9be79fde1bed786c0f
To: <sip:[email protected]>;tag=57e83ab9-6759-4d0f-a045-71829ebfc85c
Call-ID: [email protected]_1-b2b_1
CSeq: 201 BYE
Contact: Anonymous <sip:[email protected]:5060>
User-Agent: ACC
cisco-GUID: 27669653-298810312-190787488-2356164701
h323-conf-id: 27669653-298810312-190787488-2356164701
P-Asserted-Identity: <sip:+13363744673;[email protected]:5060>
P-attestation-indicator: A
P-origination-id: uuid:561517dd-c7a2-4380-84df-036b0334dec6
X-anveo-e164: 13364960050
Content-Length: 0

douglasmv · December 15, 2025, 4:07am

This is the error on the freebbx

[2025-12-14 22:53:42] NOTICE[1049504] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘sip:[email protected]’ failed for ‘185.243.5.89:64709’ (callid: 993203962-254840069-1549710129) - Failed to authenticate

Stewart1 · December 15, 2025, 5:58am

No, it’s just another attempted attack. The timestamp is 5 minutes after the call in question.

I suspect that the real problem is your router firewall rewriting the RTP source port. Can you capture WAN interface traffic to confirm that?

douglasmv · December 15, 2025, 1:22pm

douglasmv · December 15, 2025, 1:45pm

I’m trying to find this in my crdelpoint E320

Stewart1 · December 15, 2025, 2:23pm

Use tcpdump to see whether RTP is being sent to the correct IP address and port.

If so, I’d suspect the router. In some cases, forwarding Asterisk’s RTP port range (default UDP 10000-20000) to the PBX will fix it.

douglasmv · December 15, 2025, 4:54pm

Good morning, Do to the issues we were having I moved the account back to the original I had with anveo. I have ran into a new issue and that is when a caller calls in it is now playing the wrong message for that state. I made sure everything was cofigured right and when i put the DID in there it says, “Please leave a message.” Any suggestions?

douglasmv · December 15, 2025, 11:30pm

When I add the number 3364960050 to the DID, nothing works i tried to add a 1 and +1, and it says this call cannot be completed at diald

douglasmv · January 4, 2026, 10:08pm

I’m back working on the Aveno Direct. I have a test number to work on and need some help. It looks like now its finally making it to the server but still get a busy signal

douglasmv · January 4, 2026, 10:13pm

This is what I get from the Anveo Direct report.

/*>>>|174.141.213.62:5060 @ 2026-01-04 22:09:30 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK444b7a821e1bbe09b6754663f18605a5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c41173225d8a40e11f38fee9596e3b1a
To: <sip:[email protected]>
Call-ID: [email protected]_1-b2b_1
CSeq: 200 INVITE
Contact: Anonymous <sip:[email protected]:5060>
Expires: 300
User-Agent: ACC
cisco-GUID: 3658470692-170463708-221222336-1868885879
h323-conf-id: 3658470692-170463708-221222336-1868885879
P-Asserted-Identity: <sip:+13363744673;[email protected]:5060>
P-attestation-indicator: A
P-origination-id: 3f4feb19-ebfd-4423-87ea-a24d3ad4d9c0
X-anveo-e164: 16627800002
Content-Type: application/sdp
Content-Length: 281

v=0
o=Sonus_UAC 815068 698809 IN IP4 67.231.1.112
s=SIP Media Capabilities
c=IN IP4 67.231.1.80
t=0 0
m=audio 17120 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

/*>>>|174.141.213.62:5060 @ 2026-01-04 22:09:31 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK444b7a821e1bbe09b6754663f18605a5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c41173225d8a40e11f38fee9596e3b1a
To: <sip:[email protected]>
Call-ID: [email protected]_1-b2b_1
CSeq: 200 INVITE
Contact: Anonymous <sip:[email protected]:5060>
Expires: 300
User-Agent: ACC
cisco-GUID: 3658470692-170463708-221222336-1868885879
h323-conf-id: 3658470692-170463708-221222336-1868885879
P-Asserted-Identity: <sip:+13363744673;[email protected]:5060>
P-attestation-indicator: A
P-origination-id: 3f4feb19-ebfd-4423-87ea-a24d3ad4d9c0
X-anveo-e164: 16627800002
Content-Type: application/sdp
Content-Length: 281

v=0
o=Sonus_UAC 815068 698809 IN IP4 67.231.1.112
s=SIP Media Capabilities
c=IN IP4 67.231.1.80
t=0 0
m=audio 17120 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

/*>>>|174.141.213.62:5060 @ 2026-01-04 22:09:32 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK444b7a821e1bbe09b6754663f18605a5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c41173225d8a40e11f38fee9596e3b1a
To: <sip:[email protected]>
Call-ID: [email protected]_1-b2b_1
CSeq: 200 INVITE
Contact: Anonymous <sip:[email protected]:5060>
Expires: 300
User-Agent: ACC
cisco-GUID: 3658470692-170463708-221222336-1868885879
h323-conf-id: 3658470692-170463708-221222336-1868885879
P-Asserted-Identity: <sip:+13363744673;[email protected]:5060>
P-attestation-indicator: A
P-origination-id: 3f4feb19-ebfd-4423-87ea-a24d3ad4d9c0
X-anveo-e164: 16627800002
Content-Type: application/sdp
Content-Length: 281

v=0
o=Sonus_UAC 815068 698809 IN IP4 67.231.1.112
s=SIP Media Capabilities
c=IN IP4 67.231.1.80
t=0 0
m=audio 17120 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

BlazeStudios · January 4, 2026, 10:27pm

You have their IPs in your firewall and in the Match field for the trunk? Are the calls even make it to the PBX? Based on this they are sending the same INVITE 3 times because there’s no response.

douglasmv · January 4, 2026, 10:49pm

Yes I do have the match setup. As far as the firwall at this time I have it cut off for a few minutes

douglasmv · January 4, 2026, 11:31pm

douglasmv · January 4, 2026, 11:32pm

david55 · January 5, 2026, 12:16am

They’d get a response even if the match field is wrong. Either Asterisk isn’t receiving the INVITE, or its response isn’t getting through.

douglasmv · January 5, 2026, 12:26am

I know Anveo Direct is the hardest I have set up ever setup. We are setting up 50 weather lines around the USA, and this is the cheapest. The only issue is that the tech support is no help