Outbound calls not working after resore

FreePBX Providers
1

I restored my system and after I cannot make outbound calls. Inbound work. I have chek config and everthing appears correct.

Below is a trace

<— Received SIP request (1005 bytes) from UDP:192.168.0.90:64211 —>
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-da610147ca036010-1—d 8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:64211;rinstance=cb6308a5fdb81262
To: sip:[email protected]:5160
From: "60001"sip:[email protected]:5160;tag=e90c7959
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN FO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 403

v=0
o=3cxVCE 22554315 128326275 IN IP4 192.168.0.90
s=3cxVCE Audio Call
c=IN IP4 192.168.0.90
t=0 0
m=audio 40048 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40010 RTP/AVP 34
c=IN IP4 192.168.0.90
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv

<— Transmitting SIP response (559 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9h G4bK-d8754z-da610147ca036010-1—d8754z-
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
From: “60001” sip:[email protected];tag=e90c7959
To: sip:[email protected];tag=z9hG4bK-d8754z-da610147ca036010-1—d8754 z-
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1751341029/dd5d64f02f568291f039 62823dfc4011”,opaque=“1621db811e11a58a”,algorithm=MD5,qop=“auth”
Server: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Received SIP request (390 bytes) from UDP:192.168.0.90:64211 —>
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-da610147ca036010-1—d 8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=z9hG4bK-d8754z-da610147ca036010-1—d8754 z-
From: "60001"sip:[email protected]:5160;tag=e90c7959
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1301 bytes) from UDP:192.168.0.90:64211 —>
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-8d0d0778c348351d-1—d 8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:64211;rinstance=cb6308a5fdb81262
To: sip:[email protected]:5160
From: "60001"sip:[email protected]:5160;tag=e90c7959
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN FO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username=“60001”,realm=“asterisk”,nonce=“1751341029/dd5d64 f02f568291f03962823dfc4011”,uri=“sip:[email protected]:5160”,response=“44 9bc586170d09a435e89876f32b4f6b”,cnonce=“ffd38023439eb1f4b88486111e1d31ed”,nc=000 00001,qop=auth,algorithm=MD5,opaque=“1621db811e11a58a”
Content-Length: 403

v=0
o=3cxVCE 22554315 128326275 IN IP4 192.168.0.90
s=3cxVCE Audio Call
c=IN IP4 192.168.0.90
t=0 0
m=audio 40048 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40010 RTP/AVP 34
c=IN IP4 192.168.0.90
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv

<— Transmitting SIP response (359 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9h G4bK-d8754z-8d0d0778c348351d-1—d8754z-
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
From: “60001” sip:[email protected];tag=e90c7959
To: sip:[email protected]
CSeq: 2 INVITE
Server: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

[2025-06-30 20:37:09] ERROR[2326]: res_pjsip_header_funcs.c:670 remove_header: N o headers had been previously added to this session.
<— Transmitting SIP request (1079 bytes) to UDP:54.172.60.0:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 184.190.137.60:5160;rport;branch=z9hG4bKPj2f9d9234-109b-47ad-98 5c-6ef7eb1406e6
From: sip:[email protected];tag=4ca48461-1d95-40db-8b42-d74c3a4625eb
To: sip:[email protected]
Contact: sip:[email protected]:5160
Call-ID: a1cc240c-c30d-4320-b064-b93d1fcca9ed
CSeq: 26629 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NO TIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16.0.40.11(16.30.0)
Content-Type: application/sdp
Content-Length: 363

v=0
o=- 52899859 52899859 IN IP4 184.190.137.60
s=Asterisk
c=IN IP4 184.190.137.60
t=0 0
m=audio 17534 RTP/AVP 0 8 3 111 4 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Received SIP response (419 bytes) from UDP:54.172.60.0:5060 —>
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 184.190.137.60:5160;rport=5160;branch=z9hG4bKPj2f9d9234-109b-47 ad-985c-6ef7eb1406e6;received=184.190.137.60
From: sip:[email protected];tag=4ca48461-1d95-40db-8b42-d74c3a4625eb
To: sip:[email protected]
Call-ID: a1cc240c-c30d-4320-b064-b93d1fcca9ed
CSeq: 26629 INVITE
Server: Twilio Gateway
Content-Length: 0

<— Received SIP response (476 bytes) from UDP:54.172.60.0:5060 —>
SIP/2.0 403 Forbidden
CSeq: 26629 INVITE
Call-ID: a1cc240c-c30d-4320-b064-b93d1fcca9ed
From: sip:[email protected];tag=4ca48461-1d95-40db-8b42-d74c3a4625eb
To: sip:[email protected];tag=23630019_c3356d0b_ed2904d5-28 c1-49d0-a4eb-1fb275dd5d24
Via: SIP/2.0/UDP 184.190.137.60:5160;received=184.190.137.60;rport=5160;branch=z 9hG4bKPj2f9d9234-109b-47ad-985c-6ef7eb1406e6
Server: Twilio
Contact: sip:172.25.11.252:5060
Content-Length: 0

<— Transmitting SIP request (486 bytes) to UDP:54.172.60.0:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 184.190.137.60:5160;rport;branch=z9hG4bKPj2f9d9234-109b-47ad-98 5c-6ef7eb1406e6
From: sip:[email protected];tag=4ca48461-1d95-40db-8b42-d74c3a4625eb
To: sip:[email protected];tag=23630019_c3356d0b_ed2904d5-28 c1-49d0-a4eb-1fb275dd5d24
Call-ID: a1cc240c-c30d-4320-b064-b93d1fcca9ed
CSeq: 26629 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Transmitting SIP response (967 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9h G4bK-d8754z-8d0d0778c348351d-1—d8754z-
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
From: “60001” sip:[email protected];tag=e90c7959
To: sip:[email protected];tag=5bced9ff-f71a-490b-9cc7-f971fae169fe
CSeq: 2 INVITE
Server: FPBX-16.0.40.11(16.30.0)
Contact: sip:192.168.0.118:5160
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NO TIFY, PUBLISH, MESSAGE, REFER
P-Asserted-Identity: “CID:+16023627646” sip:[email protected]
Content-Type: application/sdp
Content-Length: 305

v=0
o=- 22554315 128326277 IN IP4 192.168.0.118
s=Asterisk
c=IN IP4 192.168.0.118
t=0 0
m=audio 14522 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 34

<— Transmitting SIP response (610 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9hG4bK-d8754z-8d0d0778c348351d-1—d8754z-
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
From: “60001” sip:[email protected];tag=e90c7959
To: sip:[email protected];tag=5bced9ff-f71a-490b-9cc7-f971fae169fe
CSeq: 2 INVITE
Server: FPBX-16.0.40.11(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Reason: Q.850;cause=17
P-Asserted-Identity: “CID:+16023627646” sip:[email protected]
Content-Length: 0

<— Received SIP request (383 bytes) from UDP:192.168.0.90:64211 —>
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-8d0d0778c348351d-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=5bced9ff-f71a-490b-9cc7-f971fae169fe
From: "60001"sip:[email protected]:5160;tag=e90c7959
Call-ID: ZDdlNmM4ZjJiYWJiNjRjNTg4N2QzN2QzYWFhYmU4MWQ.
CSeq: 2 ACK
Content-Length: 0

<— Received SIP request (865 bytes) from UDP:192.168.0.90:64211 —>
REGISTER sip:192.168.0.118:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-1c34ce3991060108-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:64211;rinstance=cb6308a5fdb81262
To: "60001"sip:[email protected]:5160
From: "60001"sip:[email protected]:5160;tag=3250df5a
Call-ID: ZTRmZDQ1ODVkN2Y0YzVmOTZlYTlmOTJhNzAyNzgwOGU.
CSeq: 143 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username=“60001”,realm=“asterisk”,nonce=“1751340937/f7abc8a3ce547cbb4add535ef2537704”,uri=“sip:192.168.0.118:5160”,response=“d1fd207a72c8ee164e290926a0be1ee1”,cnonce=“3dcc174bbd23561c84b27588f7465788”,nc=00000002,qop=auth,algorithm=MD5,opaque=“3509b70c190cb93e”
Content-Length: 0

<— Transmitting SIP response (577 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9hG4bK-d8754z-1c34ce3991060108-1—d8754z-
Call-ID: ZTRmZDQ1ODVkN2Y0YzVmOTZlYTlmOTJhNzAyNzgwOGU.
From: “60001” sip:[email protected];tag=3250df5a
To: “60001” sip:[email protected];tag=z9hG4bK-d8754z-1c34ce3991060108-1—d8754z-
CSeq: 143 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1751341045/d071f49b23c153289750e0acfd067d8d”,opaque=“4d110cc4654756bd”,stale=true,algorithm=MD5,qop=“auth”
Server: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Received SIP request (865 bytes) from UDP:192.168.0.90:64211 —>
REGISTER sip:192.168.0.118:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.90:64211;branch=z9hG4bK-d8754z-04281a14e82f226e-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:64211;rinstance=cb6308a5fdb81262
To: "60001"sip:[email protected]:5160
From: "60001"sip:[email protected]:5160;tag=3250df5a
Call-ID: ZTRmZDQ1ODVkN2Y0YzVmOTZlYTlmOTJhNzAyNzgwOGU.
CSeq: 144 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username=“60001”,realm=“asterisk”,nonce=“1751341045/d071f49b23c153289750e0acfd067d8d”,uri=“sip:192.168.0.118:5160”,response=“2dc5cebe6a5369cbfe7976c22679fb65”,cnonce=“e3e3186bba9c618b1561c1899c8c08a1”,nc=00000001,qop=auth,algorithm=MD5,opaque=“4d110cc4654756bd”
Content-Length: 0

<— Transmitting SIP response (541 bytes) to UDP:192.168.0.90:64211 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.90:64211;rport=64211;received=192.168.0.90;branch=z9hG4bK-d8754z-04281a14e82f226e-1—d8754z-
Call-ID: ZTRmZDQ1ODVkN2Y0YzVmOTZlYTlmOTJhNzAyNzgwOGU.
From: “60001” sip:[email protected];tag=3250df5a
To: “60001” sip:[email protected];tag=z9hG4bK-d8754z-04281a14e82f226e-1—d8754z-
CSeq: 144 REGISTER
Date: Tue, 01 Jul 2025 03:37:25 GMT
Contact: sip:[email protected]:64211;rinstance=cb6308a5fdb81262;expires=119
Expires: 120
Server: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

[2025-06-30 20:37:26] ERROR[2314]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“168.86.128.0/18”, “(null)”, …): Name or service not known
[2025-06-30 20:37:26] WARNING[2314]: acl.c:890 resolve_first: Unable to lookup ‘168.86.128.0/18’
<— Transmitting SIP request (487 bytes) to UDP:192.168.0.90:64211 —>
OPTIONS sip:[email protected]:64211;rinstance=cb6308a5fdb81262 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.118:5160;rport;branch=z9hG4bKPjc24ba4a9-0416-489b-85f7-59f639c28c77
From: sip:[email protected];tag=4f232360-72cb-426f-b560-203263628686
To: sip:[email protected];rinstance=cb6308a5fdb81262
Contact: sip:[email protected]:5160
Call-ID: 9871e6c5-03e1-46fc-9ac3-6f62288c6339
CSeq: 33891 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Received SIP response (599 bytes) from UDP:192.168.0.90:64211 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.118:5160;rport=5160;branch=z9hG4bKPjc24ba4a9-0416-489b-85f7-59f639c28c77
Contact: sip:192.168.0.90:64211
To: sip:[email protected];rinstance=cb6308a5fdb81262;tag=e74a813b
From: sip:[email protected];tag=4f232360-72cb-426f-b560-203263628686
Call-ID: 9871e6c5-03e1-46fc-9ac3-6f62288c6339
CSeq: 33891 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<— Received SIP request (899 bytes) from UDP:192.168.0.27:19272 —>
REGISTER sip:192.168.0.118:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.27:19272;rport;branch=z9hG4bK-5p4123019809364238526r
From: sip:[email protected];tag=4g7375228022448971122m
To: sip:[email protected]
Call-ID: 2e4760979139352583427k9474rmwp
CSeq: 433 REGISTER
Max-Forwards: 70
Contact: sip:[email protected]:19272
Expires: 90
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1751340901/ccdb2771aa6fc3fb59559c4ee2f0cfd4”,uri=“sip:192.168.0.118:5160”,response=“119a985be305e9030122ed6b2f1dc7bd”,opaque=“54678d68588df19d”,cnonce=“2612563647304017409”,nc=00000001,qop=auth,algorithm=MD5
User-Agent: MizuDroid/4.4.25060
Supported: replaces
Allow: ACK,PRACK,BYE,CANCEL,INVITE,UPDATE,MESSAGE,INFO,OPTIONS,SUBSCRIBE,NOTIFY,REFER
Allow-Events: presence,refer,telephone-event,keep-alive,dialog
Accept: application/sdp,application/dtmf-relay,text/plain
Content-Length: 0

[2025-06-30 20:37:42] NOTICE[2326]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘REGISTER’ from ‘sip:[email protected]’ failed for ‘192.168.0.27:19272’ (callid: 2e4760979139352583427k9474rmwp) - Failed to authenticate
<— Transmitting SIP response (522 bytes) to UDP:192.168.0.27:19272 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.27:19272;rport=19272;received=192.168.0.27;branch=z9hG4bK-5p4123019809364238526r
Call-ID: 2e4760979139352583427k9474rmwp
From: sip:[email protected];tag=4g7375228022448971122m
To: sip:[email protected];tag=z9hG4bK-5p4123019809364238526r
CSeq: 433 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1751341062/8368ee09eb48cdc60c764be06f37e3b0”,opaque=“04afeaac3a139c7e”,algorithm=MD5,qop=“auth”
Server: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Transmitting SIP request (520 bytes) to UDP:54.172.60.0:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 184.190.137.60:5160;rport;branch=z9hG4bKPjc759fbda-6c7a-4718-a2d1-8600934eae56
From: sip:[email protected];tag=dfc3a3cf-574f-401d-9356-bb86efa29f22
To: sip:[email protected]
Contact: sip:[email protected]:5160
Call-ID: 66de22b2-47d3-45fa-915f-8028f65d74ed
CSeq: 43520 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.11(16.30.0)
Content-Length: 0

<— Received SIP response (448 bytes) from UDP:54.172.60.0:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.190.137.60:5160;rport=5160;branch=z9hG4bKPjc759fbda-6c7a-4718-a2d1-8600934eae56;received=184.190.137.60
From: sip:[email protected];tag=dfc3a3cf-574f-401d-9356-bb86efa29f22
To: sip:[email protected];tag=4197692b16d247a744f8ac68815f0775.1e8b23c6
Call-ID: 66de22b2-47d3-45fa-915f-8028f65d74ed
CSeq: 43520 OPTIONS
Server: Twilio Gateway
Content-Length: 0

2

I believe that Twilio rejected the call because 480862xxxx is not a valid format for them. You need to send 1480862xxxx or +1480862xxxx.

If your old system allowed the end user to dial without the initial 1, it rewrote the number in the Outbound Route or trunk number manipulation rules.

Check to see if those got correctly restored. Also, check whether manually dialing with the initial 1 works. If you still have trouble, paste the Asterisk log (not the console output) for a failing call at pastebin.com and post the link here.

3

were do I get the astrik log? is that the one under call event login

4
5

found it. I had another trunk for internal routing witch was causing the issue. When I removed the dialing rule from that trunk and added it to the correct one it solve the issue. now my secondary issue is with DTMF not being detected on incoming calls from twilio. It is detected on internal calls. I can open another case for rthis but you guys have any sugestions?