FreePBX Anveo configuraiton on Static IP

FreePBX Providers
41

would I use the sip uri or the PSTN in the Anveo Direct setup?

42

You need to delimit the subnets by a comma. Doing a slash (/) is making it think there’s a subnet after as in x.x.x.x/28

That’s your issue, you have the Match format in an invalid format. Fix that and try again.

43

When I change the / to a comma, then it goes back to busy. I just got it to answer, but nothing was happening. I have the incoming setup but nothing.

44

That setting requires a comma delimited list of IPs and subnets (if needed).

SCR-20260104-sce
SCR-20260104-sce2548×224 23.5 KB

Using the / in between the IPs like that is invalid. Do you see any traffic hit the PBX at all? I mean at the interface level with an sngrep or grep or tshark command?

45

Yes, I’m seeing traffic. When I changed it back to the commas, everything quit. Anveo I believe it was advised to me to use the /, so I did. I left the commas now, and I’m going to keep troubleshooting.

46

This is the latest issue in freepbx

[2026-01-04 21:29:47] WARNING[9508] res_pjsip_outbound_registration.c: No response received from ‘sip:sip.anveo.com:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘60’

47

but im only receving calls not making.

48

You need to provide a PCAP/debug of an attempt to make an outbound call that fails.

49

This is from Aveno Direct

/*>>>|174.141.213.62:5060 @ 2026-01-05 03:16:10 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK4f7f7c4bbb1c6d6d411e54a6dcf01f13;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
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: 757394277-3111315197-1337555049-3129519233
h323-conf-id: 757394277-3111315197-1337555049-3129519233
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 553213 198335 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.5.85
t=0 0
m=audio 42096 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 @ 2026-01-05 03:16:11 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK4f7f7c4bbb1c6d6d411e54a6dcf01f13
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
To: <sip:[email protected]:5060>
CSeq: 200 INVITE
Server: FPBX-17.0.24(22.6.0)
Content-Length:  0


/*<<<|174.141.213.62:5060 @ 2026-01-05 03:16:11 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK4f7f7c4bbb1c6d6d411e54a6dcf01f13
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
To: <sip:[email protected]:5060>;tag=bdb5c678-00ec-4eed-ad25-2a464a912d55
CSeq: 200 INVITE
Server: FPBX-17.0.24(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:   233

v=0
o=- 553213 198337 IN IP4 174.141.213.62
s=Asterisk
c=IN IP4 174.141.213.62
t=0 0
m=audio 13712 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

/*>>>|174.141.213.62:5060 @ 2026-01-05 03:16:11 */
ACK sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;rport;branch=z9hG4bKcab38e7f3fedf0036b16d4c64a18a211
Max-Forwards: 70
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
To: <sip:[email protected]>;tag=bdb5c678-00ec-4eed-ad25-2a464a912d55
Call-ID: [email protected]_1-b2b_1
CSeq: 200 ACK
User-Agent: ACC
Content-Length: 0


/*>>>|174.141.213.62:5060 @ 2026-01-05 03:16:15 */
BYE sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK9a65b61b73142fd9bd43725839014736;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
To: <sip:[email protected]>;tag=bdb5c678-00ec-4eed-ad25-2a464a912d55
Call-ID: [email protected]_1-b2b_1
CSeq: 201 BYE
Contact: Anonymous <sip:[email protected]:5060>
User-Agent: ACC
cisco-GUID: 757394277-3111315197-1337555049-3129519233
h323-conf-id: 757394277-3111315197-1337555049-3129519233
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-Length: 0


/*<<<|174.141.213.62:5060 @ 2026-01-05 03:16:15 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK9a65b61b73142fd9bd43725839014736
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=c94de95a7d03a0e8d503584b7c2f7c91
To: <sip:[email protected]:5060>;tag=bdb5c678-00ec-4eed-ad25-2a464a912d55
CSeq: 201 BYE
Server: FPBX-17.0.24(22.6.0)
Content-Length:  0
50

This is Freepbx

[2026-01-04 22:33:09] VERBOSE[9855] res_pjsip_logger.c: <— Transmitting SIP request (448 bytes) to UDP:169.48.232.158:5060 —>

5312OPTIONS sip:sbc.anveo.com:5060 SIP/2.0

5313Via: SIP/2.0/UDP 174.141.213.62:5060;rport;branch=z9hG4bKPje9ed83ce-a61a-4682-94ae-4cd4d66806a1

5314From: sip:[email protected];tag=148f9fc9-4b4c-48cd-8063-d209680bc620

5315To: sip:sbc.anveo.com

5316Contact: sip:Bro.\[email protected]:5060

5317Call-ID: 26ec2cfa-9e8b-46e6-bfca-d61f3ca60c79

5318CSeq: 8373 OPTIONS

5319Max-Forwards: 70

5320User-Agent: FPBX-17.0.24(22.6.0)

5321Content-Length: 0

5322

5323

5324[2026-01-04 22:33:09] VERBOSE[9854] res_pjsip_logger.c: <— Received SIP response (372 bytes) from UDP:169.48.232.158:5060 —>

5325SIP/2.0 200 OK

5326Via: SIP/2.0/UDP 192.168.0.26:5060;rport=5060;branch=z9hG4bKPje9ed83ce-a61a-4682-94ae-4cd4d66806a1

5327From: sip:Bro.\[email protected]:5060;tag=148f9fc9-4b4c-48cd-8063-d209680bc620

5328To: sip:sbc.anveo.com;tag=9c20e782a1815135a6c4af546f9cae0f

5329Call-ID: 26ec2cfa-9e8b-46e6-bfca-d61f3ca60c79

5330CSeq: 8373 OPTIONS

5331Server: ACC

5332Content-Length: 0

51

Your first log is a successful call. Your second log is a successful qualify. They are not related to each other.

52

You’ve previously shown Anveo sending to Asterisk, which indicates that either Asteirsk has registered to it, or that no registration is needed. In the former case, the response is not getting through.

53

they show as successful but when I call it rings once then goes bussy

54

The problem is I have a regular anveo account and a anveo direct account but the setup on the anveo is totally different since it uses all ip addresses.

55

I think there is an issue here I have it so the authorixation is turned off but its still saying its sending a registration.

/*>>>|174.141.213.62:5060 @ 2026-01-08 21:01:24 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK00024162ce382fb5b2fc2892f72d1536;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=f3ff7e7997fefc8fabbd2bd433141782
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: 1516184504-2324479491-2442521896-2137720156
h323-conf-id: 1516184504-2324479491-2442521896-2137720156
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 769524 545084 IN IP4 67.231.1.112
s=SIP Media Capabilities
c=IN IP4 67.231.1.80
t=0 0
m=audio 52116 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 @ 2026-01-08 21:01:25 */
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK00024162ce382fb5b2fc2892f72d1536
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=f3ff7e7997fefc8fabbd2bd433141782
To: <sip:[email protected]:5060>;tag=z9hG4bK00024162ce382fb5b2fc2892f72d1536
CSeq: 200 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1767906085/24c53fdbcea4d44f66fda5a115cb8364",opaque="7bae843f5268b60c",algorithm=MD5,qop="auth"
Server: FPBX-17.0.24(22.6.0)
Content-Length:  0


/*>>>|174.141.213.62:5060 @ 2026-01-08 21:01:25 */
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;rport;branch=z9hG4bK00024162ce382fb5b2fc2892f72d1536
Max-Forwards: 70
From: <sip:[email protected]>;tag=f3ff7e7997fefc8fabbd2bd433141782
To: <sip:[email protected]>;tag=z9hG4bK00024162ce382fb5b2fc2892f72d1536
Call-ID: [email protected]_1-b2b_1
CSeq: 200 ACK
User-Agent: ACC
Content-Length: 0
56

Well it looks like i’m getting somwere but im still not gettng oudio. I opend up the 10,000-20,000 UDP and not sure what els to do.

57

My latest sip trace from Anveo Direct

58 
/*>>>|174.141.213.62:5060 @ 2026-01-09 01:10:23 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bKc1f56be0ed5e46616e794294259202d9;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
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: 1360931501-2023074877-2035223128-1328122506
h323-conf-id: 1360931501-2023074877-2035223128-1328122506
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: 279

v=0
o=Sonus_UAC 750358 859872 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.4.7
t=0 0
m=audio 2910 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 @ 2026-01-09 01:10:23 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bKc1f56be0ed5e46616e794294259202d9
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
To: <sip:[email protected]:5060>
CSeq: 200 INVITE
Server: FPBX-17.0.24(22.6.0)
Content-Length:  0


/*<<<|174.141.213.62:5060 @ 2026-01-09 01:10:23 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bKc1f56be0ed5e46616e794294259202d9
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
To: <sip:[email protected]:5060>;tag=241118b1-7b36-498f-8f24-cf832aa2624b
CSeq: 200 INVITE
Server: FPBX-17.0.24(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=- 750358 859874 IN IP4 174.141.213.62
s=Asterisk
c=IN IP4 174.141.213.62
t=0 0
m=audio 17400 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 @ 2026-01-09 01:10:23 */
ACK sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;rport;branch=z9hG4bK527255489a57fd44d8678ff9f1bac85f
Max-Forwards: 70
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
To: <sip:[email protected]>;tag=241118b1-7b36-498f-8f24-cf832aa2624b
Call-ID: [email protected]_1-b2b_1
CSeq: 200 ACK
User-Agent: ACC
Content-Length: 0


/*>>>|174.141.213.62:5060 @ 2026-01-09 01:10:30 */
BYE sip:174.141.213.62:5060 SIP/2.0
Via: SIP/2.0/UDP 169.48.232.158:5060;branch=z9hG4bK3288dcdd1b01948cb733c62d739cb5d2;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
To: <sip:[email protected]>;tag=241118b1-7b36-498f-8f24-cf832aa2624b
Call-ID: [email protected]_1-b2b_1
CSeq: 201 BYE
Contact: Anonymous <sip:[email protected]:5060>
User-Agent: ACC
cisco-GUID: 1360931501-2023074877-2035223128-1328122506
h323-conf-id: 1360931501-2023074877-2035223128-1328122506
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-Length: 0


/*<<<|174.141.213.62:5060 @ 2026-01-09 01:10:30 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 169.48.232.158:5060;rport=5060;received=169.48.232.158;branch=z9hG4bK3288dcdd1b01948cb733c62d739cb5d2
Call-ID: [email protected]_1-b2b_1
From: <sip:[email protected]>;tag=74762a10b35e5cd3c5076d9574348aba
To: <sip:[email protected]:5060>;tag=241118b1-7b36-498f-8f24-cf832aa2624b
CSeq: 201 BYE
Server: FPBX-17.0.24(22.6.0)
Content-Length:  0
59

There is no reference to registration in your log fragment. Registration is not designed for authentication. It uses it, but is not part of it.

60

Confirm that you forwarded UDP ports 10000-20000 from any source address to the LAN IP of the PBX.

For testing, route the incoming call to a working extension (use *43 echo test to confirm) and report on the presence of both incoming and outgoing audio.

Also, report whether in and out audio is present on outgoing calls.

Router/firewall make/model? Does it have your public IP on its WAN interface? What VoIP related settings do you have?