PBX down, strange authorization problem

General Help
1

Our main PBX is in trouble, and I can not figure it out. The PBX has been running for years, and no changes have been made recently.

Yesterday, my provider Frontier gave me a new router. I installed it and configured it, but was not happy with the NAT. So, I went back to the old router. Nothing was changed in the settings!

Everything works fine on our network, but the PBX does many strange things.

Fail2ban blocks login attempts from my phones despite them being in the whitelist. Even my Sipstation trunks showed up in the blocked list.

Now I have Fail2ban off, and the phones register and deregister often.

Every call (internally or through sipstation) gets rejected:

rejected because extension not found in context ‘from-pstn’.
rejected because extension not found in context ‘from-internal’.

I have reloaded, rebooted, etc. I have been on this for hours, and I ran out of ideas.

Any hints?

2

Turn off SIP ALG on the router.

Look at the request URI in the incoming INVITEs and try and work out what it actually represents, and why (at least for the extensions) it isn’t the dialled number.

3

Thank you. No SIP ALG setting on the router. Old Fios router.

Where can I see the INVITEs? They look okay on the phone.

4

In the full log after using “pjsip set logger on” via the CLI (or “sip set debug on”, if you are in need of an upgrade).

5

Thank you! The invite looks okay to me.

9602	<--- SIP read from UDP:192.168.1.112:5060 --->	
9603	INVITE sip:[email protected];user=phone SIP/2.0	
9604	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;rport	
9605	From: "Property" <sip:[email protected]>;tag=3671815548	
9606	To: <sip:[email protected];user=phone>	
9607	Call-ID: 2500745707@192_168_1_112	
9608	CSeq: 2 INVITE	
9609	Contact: <sip:[email protected]:5060>	
9610	Max-Forwards: 70	
9611	User-Agent: C470IP/022270000000	
9612	Supported: replaces	
9613	Allow-Events: message-summary, refer	
9614	Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY	
9615	Content-Type: application/sdp	
9616	Content-Length: 370	
9617		
9618	v=0	
9619	o=4611 5008 7 IN IP4 192.168.1.112	
9620	s=Mapping	
9621	c=IN IP4 192.168.1.112	
9622	t=0 0	
9623	m=audio 5008 RTP/AVP 9 8 0 96 97 2 18 101	
9624	a=rtpmap:9 G722/8000	
9625	a=rtpmap:8 PCMA/8000	
9626	a=rtpmap:0 PCMU/8000	
9627	a=rtpmap:96 G726-32/8000	
9628	a=rtpmap:97 AAL2-G726-32/8000	
9629	a=rtpmap:2 G726-32/8000	
9630	a=rtpmap:18 G729/8000	
9631	a=fmtp:18 annexb=no	
9632	a=rtpmap:101 telephone-event/8000	
9633	a=fmtp:101 0-16	
9634	<------------->	
9635	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (14 headers 16 lines) ---	
9636	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: Sending to 192.168.1.112:5060 (NAT)	
9637	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Sending to 192.168.1.112:5060 (NAT)	
9638	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Using INVITE request as basis request - 2500745707@192_168_1_112	
9639	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found peer '4611' for '4611' from 192.168.1.112:5060	
9640	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c:	
9641	<--- Reliably Transmitting (no NAT) to 192.168.1.112:5060 --->	
9642	SIP/2.0 401 Unauthorized	
9643	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;received=192.168.1.112;rport=5060	
9644	From: "Property" <sip:[email protected]>;tag=3671815548	
9645	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9646	Call-ID: 2500745707@192_168_1_112	
9647	CSeq: 2 INVITE	
9648	Server: FPBX-15.0.37.4(16.30.0)	
9649	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9650	Supported: replaces, timer	
9651	WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5fa569f9"	
9652	Content-Length: 0	
9653		
9654		
9655	<------------>	
9656	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Scheduling destruction of SIP dialog '2500745707@192_168_1_112' in 7616 ms (Method: INVITE)	
9657	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c:	
9658	<--- SIP read from UDP:192.168.1.112:5060 --->	
9659	ACK sip:[email protected];user=phone SIP/2.0	
9660	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;rport	
9661	From: "Property" <sip:[email protected]>;tag=3671815548	
9662	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9663	Call-ID: 2500745707@192_168_1_112	
9664	CSeq: 2 ACK	
9665	Contact: <sip:[email protected]:5060>	
9666	Max-Forwards: 70	
9667	User-Agent: C470IP/022270000000	
9668	Content-Length: 0	
9669		
9670	<------------->	
9671	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (10 headers 0 lines) ---	
9672	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c:	
9673	<--- SIP read from UDP:192.168.1.112:5060 --->	
9674	INVITE sip:[email protected];user=phone SIP/2.0	
9675	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;rport	
9676	From: "Property" <sip:[email protected]>;tag=3671815548	
9677	To: <sip:[email protected];user=phone>	
9678	Call-ID: 2500745707@192_168_1_112	
9679	CSeq: 3 INVITE	
9680	Contact: <sip:[email protected]:5060>	
9681	Authorization: Digest username="4611", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="5fa569f9", response="9e2da9473dff1cb8a644de7d701229db"	
9682	Max-Forwards: 70	
9683	User-Agent: C470IP/022270000000	
9684	Supported: replaces	
9685	Allow-Events: message-summary, refer	
9686	Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY	
9687	Content-Type: application/sdp	
9688	Content-Length: 370	
9689		
9690	v=0	
9691	o=4611 5008 7 IN IP4 192.168.1.112	
9692	s=Mapping	
9693	c=IN IP4 192.168.1.112	
9694	t=0 0	
9695	m=audio 5008 RTP/AVP 9 8 0 96 97 2 18 101	
9696	a=rtpmap:9 G722/8000	
9697	a=rtpmap:8 PCMA/8000	
9698	a=rtpmap:0 PCMU/8000	
9699	a=rtpmap:96 G726-32/8000	
9700	a=rtpmap:97 AAL2-G726-32/8000	
9701	a=rtpmap:2 G726-32/8000	
9702	a=rtpmap:18 G729/8000	
9703	a=fmtp:18 annexb=no	
9704	a=rtpmap:101 telephone-event/8000	
9705	a=fmtp:101 0-16	
9706	<------------->	
9707	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (15 headers 16 lines) ---	
9708	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Sending to 192.168.1.112:5060 (no NAT)	
9709	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Using INVITE request as basis request - 2500745707@192_168_1_112	
9710	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found peer '4611' for '4611' from 192.168.1.112:5060	
9711	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] netsock2.c: Using SIP RTP TOS bits 184	
9712	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] netsock2.c: Using SIP RTP CoS mark 5	
9713	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Got SDP version 7 and unique parts [4611 5008 IN IP4 192.168.1.112]	
9714	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 9	
9715	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 8	
9716	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 0	
9717	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 96	
9718	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 97	
9719	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 2	
9720	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 18	
9721	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 101	
9722	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G722 for ID 9	
9723	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format PCMA for ID 8	
9724	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format PCMU for ID 0	
9725	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G726-32 for ID 96	
9726	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format AAL2-G726-32 for ID 97	
9727	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G726-32 for ID 2	
9728	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G729 for ID 18	
9729	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format telephone-event for ID 101	
9730	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729|g726|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)	
9731	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)	
9732	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Peer audio RTP is at port 192.168.1.112:5008	
9733	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Looking for xxxxxxxxxx in from-internal (domain 192.168.1.151)	
9734	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c:	
9735	<--- Reliably Transmitting (no NAT) to 192.168.1.112:5060 --->	
9736	SIP/2.0 404 Not Found	
9737	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;received=192.168.1.112;rport=5060	
9738	From: "Property" <sip:[email protected]>;tag=3671815548	
9739	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9740	Call-ID: 2500745707@192_168_1_112	
9741	CSeq: 3 INVITE	
9742	Server: FPBX-15.0.37.4(16.30.0)	
9743	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9744	Supported: replaces, timer	
9745	Content-Length: 0	
9746		
9747		
9748	<------------>	
9749	[2024-02-14 12:21:12] NOTICE[12233][C-00000012] chan_sip.c: Call from '4611' (192.168.1.112:5060) to extension 'xxxxxxxxxx' rejected because extension not found in context 'from-internal'.	
9750	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Scheduling destruction of SIP dialog '2500745707@192_168_1_112' in 7616 ms (Method: INVITE)	
9751	[2024-02-14 12:21:13] VERBOSE[12233] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.1.112:5060:	
9752	SIP/2.0 404 Not Found	
9753	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;received=192.168.1.112;rport=5060	
9754	From: "Property" <sip:[email protected]>;tag=3671815548	
9755	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9756	Call-ID: 2500745707@192_168_1_112	
9757	CSeq: 3 INVITE	
9758	Server: FPBX-15.0.37.4(16.30.0)	
9759	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9760	Supported: replaces, timer	
9761	Content-Length: 0	
9762		
9763		
9764	---	
9765	[2024-02-14 12:21:13] VERBOSE[12233] chan_sip.c:	
9766	<--- SIP read from UDP:192.168.1.112:5060 --->	
9767	ACK sip:[email protected];user=phone SIP/2.0	
9768	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;rport	
9769	From: "Property" <sip:[email protected]>;tag=3671815548	
9770	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9771	Call-ID: 2500745707@192_168_1_112	
9772	CSeq: 3 ACK	
9773	Contact: <sip:[email protected]:5060>	
9774	Authorization: Digest username="4611", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="5fa569f9", response="9e2da9473dff1cb8a644de7d701229db"	
9775	Max-Forwards: 70	
9776	User-Agent: C470IP/022270000000	
9777	Content-Length: 0	
9778		
9779	<------------->9600	[2024-02-14 12:21:11] VERBOSE[12233] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: SUBSCRIBE	
9601	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c:	
9602	<--- SIP read from UDP:192.168.1.112:5060 --->	
9603	INVITE sip:[email protected];user=phone SIP/2.0	
9604	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;rport	
9605	From: "Property" <sip:[email protected]>;tag=3671815548	
9606	To: <sip:[email protected];user=phone>	
9607	Call-ID: 2500745707@192_168_1_112	
9608	CSeq: 2 INVITE	
9609	Contact: <sip:[email protected]:5060>	
9610	Max-Forwards: 70	
9611	User-Agent: C470IP/022270000000	
9612	Supported: replaces	
9613	Allow-Events: message-summary, refer	
9614	Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY	
9615	Content-Type: application/sdp	
9616	Content-Length: 370	
9617		
9618	v=0	
9619	o=4611 5008 7 IN IP4 192.168.1.112	
9620	s=Mapping	
9621	c=IN IP4 192.168.1.112	
9622	t=0 0	
9623	m=audio 5008 RTP/AVP 9 8 0 96 97 2 18 101	
9624	a=rtpmap:9 G722/8000	
9625	a=rtpmap:8 PCMA/8000	
9626	a=rtpmap:0 PCMU/8000	
9627	a=rtpmap:96 G726-32/8000	
9628	a=rtpmap:97 AAL2-G726-32/8000	
9629	a=rtpmap:2 G726-32/8000	
9630	a=rtpmap:18 G729/8000	
9631	a=fmtp:18 annexb=no	
9632	a=rtpmap:101 telephone-event/8000	
9633	a=fmtp:101 0-16	
9634	<------------->	
9635	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (14 headers 16 lines) ---	
9636	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: Sending to 192.168.1.112:5060 (NAT)	
9637	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Sending to 192.168.1.112:5060 (NAT)	
9638	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Using INVITE request as basis request - 2500745707@192_168_1_112	
9639	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found peer '4611' for '4611' from 192.168.1.112:5060	
9640	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c:	
9641	<--- Reliably Transmitting (no NAT) to 192.168.1.112:5060 --->	
9642	SIP/2.0 401 Unauthorized	
9643	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;received=192.168.1.112;rport=5060	
9644	From: "Property" <sip:[email protected]>;tag=3671815548	
9645	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9646	Call-ID: 2500745707@192_168_1_112	
9647	CSeq: 2 INVITE	
9648	Server: FPBX-15.0.37.4(16.30.0)	
9649	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9650	Supported: replaces, timer	
9651	WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5fa569f9"	
9652	Content-Length: 0	
9653		
9654		
9655	<------------>	
9656	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Scheduling destruction of SIP dialog '2500745707@192_168_1_112' in 7616 ms (Method: INVITE)	
9657	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c:	
9658	<--- SIP read from UDP:192.168.1.112:5060 --->	
9659	ACK sip:[email protected];user=phone SIP/2.0	
9660	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK21b3eb131532e4b547778cc6a1534f92;rport	
9661	From: "Property" <sip:[email protected]>;tag=3671815548	
9662	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9663	Call-ID: 2500745707@192_168_1_112	
9664	CSeq: 2 ACK	
9665	Contact: <sip:[email protected]:5060>	
9666	Max-Forwards: 70	
9667	User-Agent: C470IP/022270000000	
9668	Content-Length: 0	
9669		
9670	<------------->	
9671	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (10 headers 0 lines) ---	
9672	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c:	
9673	<--- SIP read from UDP:192.168.1.112:5060 --->	
9674	INVITE sip:[email protected];user=phone SIP/2.0	
9675	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;rport	
9676	From: "Property" <sip:[email protected]>;tag=3671815548	
9677	To: <sip:[email protected];user=phone>	
9678	Call-ID: 2500745707@192_168_1_112	
9679	CSeq: 3 INVITE	
9680	Contact: <sip:[email protected]:5060>	
9681	Authorization: Digest username="4611", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="5fa569f9", response="9e2da9473dff1cb8a644de7d701229db"	
9682	Max-Forwards: 70	
9683	User-Agent: C470IP/022270000000	
9684	Supported: replaces	
9685	Allow-Events: message-summary, refer	
9686	Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY	
9687	Content-Type: application/sdp	
9688	Content-Length: 370	
9689		
9690	v=0	
9691	o=4611 5008 7 IN IP4 192.168.1.112	
9692	s=Mapping	
9693	c=IN IP4 192.168.1.112	
9694	t=0 0	
9695	m=audio 5008 RTP/AVP 9 8 0 96 97 2 18 101	
9696	a=rtpmap:9 G722/8000	
9697	a=rtpmap:8 PCMA/8000	
9698	a=rtpmap:0 PCMU/8000	
9699	a=rtpmap:96 G726-32/8000	
9700	a=rtpmap:97 AAL2-G726-32/8000	
9701	a=rtpmap:2 G726-32/8000	
9702	a=rtpmap:18 G729/8000	
9703	a=fmtp:18 annexb=no	
9704	a=rtpmap:101 telephone-event/8000	
9705	a=fmtp:101 0-16	
9706	<------------->	
9707	[2024-02-14 12:21:12] VERBOSE[12233] chan_sip.c: --- (15 headers 16 lines) ---	
9708	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Sending to 192.168.1.112:5060 (no NAT)	
9709	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Using INVITE request as basis request - 2500745707@192_168_1_112	
9710	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found peer '4611' for '4611' from 192.168.1.112:5060	
9711	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] netsock2.c: Using SIP RTP TOS bits 184	
9712	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] netsock2.c: Using SIP RTP CoS mark 5	
9713	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Got SDP version 7 and unique parts [4611 5008 IN IP4 192.168.1.112]	
9714	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 9	
9715	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 8	
9716	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 0	
9717	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 96	
9718	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 97	
9719	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 2	
9720	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 18	
9721	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found RTP audio format 101	
9722	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G722 for ID 9	
9723	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format PCMA for ID 8	
9724	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format PCMU for ID 0	
9725	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G726-32 for ID 96	
9726	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format AAL2-G726-32 for ID 97	
9727	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G726-32 for ID 2	
9728	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format G729 for ID 18	
9729	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Found audio description format telephone-event for ID 101	
9730	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729|g726|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)	
9731	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)	
9732	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Peer audio RTP is at port 192.168.1.112:5008	
9733	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Looking for xxxxxxxxxx in from-internal (domain 192.168.1.151)	
9734	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c:	
9735	<--- Reliably Transmitting (no NAT) to 192.168.1.112:5060 --->	
9736	SIP/2.0 404 Not Found	
9737	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;received=192.168.1.112;rport=5060	
9738	From: "Property" <sip:[email protected]>;tag=3671815548	
9739	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9740	Call-ID: 2500745707@192_168_1_112	
9741	CSeq: 3 INVITE	
9742	Server: FPBX-15.0.37.4(16.30.0)	
9743	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9744	Supported: replaces, timer	
9745	Content-Length: 0	
9746		
9747		
9748	<------------>	
9749	[2024-02-14 12:21:12] NOTICE[12233][C-00000012] chan_sip.c: Call from '4611' (192.168.1.112:5060) to extension 'xxxxxxxxxx' rejected because extension not found in context 'from-internal'.	
9750	[2024-02-14 12:21:12] VERBOSE[12233][C-00000012] chan_sip.c: Scheduling destruction of SIP dialog '2500745707@192_168_1_112' in 7616 ms (Method: INVITE)	
9751	[2024-02-14 12:21:13] VERBOSE[12233] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.1.112:5060:	
9752	SIP/2.0 404 Not Found	
9753	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;received=192.168.1.112;rport=5060	
9754	From: "Property" <sip:[email protected]>;tag=3671815548	
9755	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9756	Call-ID: 2500745707@192_168_1_112	
9757	CSeq: 3 INVITE	
9758	Server: FPBX-15.0.37.4(16.30.0)	
9759	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
9760	Supported: replaces, timer	
9761	Content-Length: 0	
9762		
9763		
9764	---	
9765	[2024-02-14 12:21:13] VERBOSE[12233] chan_sip.c:	
9766	<--- SIP read from UDP:192.168.1.112:5060 --->	
9767	ACK sip:[email protected];user=phone SIP/2.0	
9768	Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK493b58982382a06699106a3b294bc48a;rport	
9769	From: "Property" <sip:[email protected]>;tag=3671815548	
9770	To: <sip:[email protected];user=phone>;tag=as08eadd21	
9771	Call-ID: 2500745707@192_168_1_112	
9772	CSeq: 3 ACK	
9773	Contact: <sip:[email protected]:5060>	
9774	Authorization: Digest username="4611", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="5fa569f9", response="9e2da9473dff1cb8a644de7d701229db"	
9775	Max-Forwards: 70	
9776	User-Agent: C470IP/022270000000	
9777	Content-Length: 0	
9778		
9779	<------------->
6

As this is all on the local network, it can not have anything to do with the router or internet connection.

No upgrades have been made in the last two weeks. Yesterday, everything was running fine.

What is going on?

7

It seems like something problematic was introduced to your network and that is where you’ll maybe need to look. You haven’t really provided any relevant information that would easily explain what you are experiencing here.

All symptoms you are mentioning here would be symptoms of a broader network issue and that’s where you’ll need to keep looking.

8

Okay, I believe I can rule out that it is a network problem. All my network tests, even with other VOIP tests, work 100% fine.

Any INVITE comes in without authorization. Now authorization is requested. The communication back and forth works fine. Another INVITE is sent with authorization, and CSeq is incremented by 1. The peer is found but rejected as unauthorized.

The passwords are correct.

9

This is from outside with NAT (The first sample was on the local network only, no NAT)

	13:50:02] VERBOSE[27885] chan_sip.c:	
	275583	<--- SIP read from TLS:166.172.188.129:61834 --->	
	275584	INVITE sip:[email protected]:5061 SIP/2.0	
	275585	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKe0PTnwAXHZiFgY83;rport	
	275586	Contact: <sip:[email protected]:61834;transport=tls>;video	
	275587	Max-Forwards: 70	
	275588	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275589	Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER	
	275590	Supported: replaces, path	
	275591	To: <sip:[email protected]:5061>	
	275592	Content-Type: application/sdp	
	275593	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275594	CSeq: 1 INVITE	
	275595	User-Agent: Groundwire/5.9.68 (build 2064866; iOS 17.2.1; arm64-neon)	
	275596	Content-Length: 386	
	275597		
	275598	v=0	
	275599	o=- 3351099496 21367 IN IP4 172.26.170.170	
	275600	s=hkeskit	
	275601	c=IN IP4 10.93.135.225	
	275602	t=0 0	
	275603	m=audio 15622 RTP/AVP 18 103 102 3 0 8 9 101	
	275604	a=rtpmap:101 telephone-event/8000	
	275605	a=rtpmap:102 iLBC/8000	
	275606	a=rtpmap:103 opus/48000/2	
	275607	a=fmtp:101 0-15	
	275608	a=fmtp:102 mode=30	
	275609	a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1	
	275610	a=fmtp:18 annexb=no	
	275611	a=ptime:30	
	275612	a=sendrecv	
	275613	<------------->	
	275614	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c: --- (13 headers 15 lines) ---	
	275615	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c: Sending to 166.172.188.129:61834 (NAT)	
	275616	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Sending to 166.172.188.129:61834 (NAT)	
	275617	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Using INVITE request as basis request - 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275618	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Found peer '4702' for '4702' from 166.172.188.129:61834	
	275619	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c:	
	275620	<--- Reliably Transmitting (NAT) to 166.172.188.129:61834 --->	
	275621	SIP/2.0 401 Unauthorized	
	275622	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKe0PTnwAXHZiFgY83;received=166.172.188.129;rport=61834	
	275623	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275624	To: <sip:[email protected]:5061>;tag=as65b9dc4d	
	275625	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275626	CSeq: 1 INVITE	
	275627	Server: FPBX-15.0.37.4(16.30.0)	
	275628	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
	275629	Supported: replaces, timer	
	275630	WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5283d25e"	
	275631	Content-Length: 0	
	275632		
	275633		
	275634	<------------>	
	275635	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Scheduling destruction of SIP dialog '7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200' in 6400 ms (Method: INVITE)	
	275636	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c:	
	275637	<--- SIP read from TLS:166.172.188.129:61834 --->	
	275638	ACK sip:[email protected]:5061 SIP/2.0	
	275639	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKe0PTnwAXHZiFgY83;rport	
	275640	Max-Forwards: 70	
	275641	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275642	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275643	To: <sip:[email protected]:5061>;tag=as65b9dc4d	
	275644	CSeq: 1 ACK	
	275645	User-Agent: Groundwire/5.9.68 (build 2064866; iOS 17.2.1; arm64-neon)	
	275646	Content-Length: 0	
	275647		
	275648	<------------->	
	275649	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c: --- (9 headers 0 lines) ---	
	275650	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c:	
	275651	<--- SIP read from TLS:166.172.188.129:61834 --->	
	275652	INVITE sip:[email protected]:5061 SIP/2.0	
	275653	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKW20PeV72PgmbUv6S;rport	
	275654	Contact: <sip:[email protected]:61834;transport=tls>;video	
	275655	Max-Forwards: 70	
	275656	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275657	Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER	
	275658	Supported: replaces, path	
	275659	To: <sip:[email protected]:5061>	
	275660	Content-Type: application/sdp	
	275661	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275662	CSeq: 2 INVITE	
	275663	Authorization: Digest username="4702",realm="asterisk",algorithm=MD5,uri="sip:[email protected]:5061",nonce="5283d25e",response="12b48ca86433236af4bf3ec123bfa610"	
	275664	User-Agent: Groundwire/5.9.68 (build 2064866; iOS 17.2.1; arm64-neon)	
	275665	Content-Length: 386	
	275666		
	275667	v=0	
	275668	o=- 3351099496 21367 IN IP4 172.26.170.170	
	275669	s=hkeskit	
	275670	c=IN IP4 10.93.135.225	
	275671	t=0 0	
	275672	m=audio 15622 RTP/AVP 18 103 102 3 0 8 9 101	
	275673	a=rtpmap:101 telephone-event/8000	
	275674	a=rtpmap:102 iLBC/8000	
	275675	a=rtpmap:103 opus/48000/2	
	275676	a=fmtp:101 0-15	
	275677	a=fmtp:102 mode=30	
	275678	a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1	
	275679	a=fmtp:18 annexb=no	
	275680	a=ptime:30	
	275681	a=sendrecv	
	275682	<------------->	
	275683	[2024-02-14 13:50:02] VERBOSE[27885] chan_sip.c: --- (14 headers 15 lines) ---	
	275684	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Sending to 166.172.188.129:61834 (NAT)	
	275685	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Using INVITE request as basis request - 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275686	[2024-02-14 13:50:02] VERBOSE[27885][C-0000001c] chan_sip.c: Found peer '4702' for '4702' from 166.172.188.129:61834	
	275687	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] netsock2.c: Using SIP RTP TOS bits 184	
	275688	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] netsock2.c: Using SIP RTP CoS mark 5	
	275689	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Got SDP version 21367 and unique parts [- 3351099496 IN IP4 172.26.170.170]	
	275690	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 18	
	275691	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 103	
	275692	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 102	
	275693	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 3	
	275694	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 0	
	275695	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 8	
	275696	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 9	
	275697	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found RTP audio format 101	
	275698	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found audio description format telephone-event for ID 101	
	275699	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found audio description format iLBC for ID 102	
	275700	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Found audio description format opus for ID 103	
	275701	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|g729|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)	
	275702	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)	
	275703	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Peer audio RTP is at port 10.93.135.225:15622	
	275704	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Looking for xxxxxxxxxx in from-internal (domain mypbxdomain.com)	
	275705	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c:	
	275706	<--- Reliably Transmitting (NAT) to 166.172.188.129:61834 --->	
	275707	SIP/2.0 404 Not Found	
	275708	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKW20PeV72PgmbUv6S;received=166.172.188.129;rport=61834	
	275709	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275710	To: <sip:[email protected]:5061>;tag=as65b9dc4d	
	275711	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275712	CSeq: 2 INVITE	
	275713	Server: FPBX-15.0.37.4(16.30.0)	
	275714	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE	
	275715	Supported: replaces, timer	
	275716	Content-Length: 0	
	275717		
	275718		
	275719	<------------>	
	275720	[2024-02-14 13:50:03] NOTICE[27885][C-0000001c] chan_sip.c: Call from '4702' (166.172.188.129:61834) to extension 'xxxxxxxxxx' rejected because extension not found in context 'from-internal'.	
	275721	[2024-02-14 13:50:03] VERBOSE[27885][C-0000001c] chan_sip.c: Scheduling destruction of SIP dialog '7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200' in 6400 ms (Method: INVITE)	
	275722	[2024-02-14 13:50:03] VERBOSE[12233] chan_sip.c: Really destroying SIP dialog '1857145116@192_168_1_112' Method: REGISTER	
	275723	[2024-02-14 13:50:03] VERBOSE[27885] chan_sip.c:	
	275724	<--- SIP read from TLS:166.172.188.129:61834 --->	
	275725	ACK sip:[email protected]:5061 SIP/2.0	
	275726	Via: SIP/2.0/TLS 10.93.135.225:55001;branch=z9hG4bKW20PeV72PgmbUv6S;rport	
	275727	Max-Forwards: 70	
	275728	Call-ID: 7BEE41216C49C8B9B27F4C57C0E5A8061CC1B200	
	275729	From: <sip:[email protected]:5061>;tag=56F1AE46A35EE96220FDD53563EE51D8	
	275730	To: <sip:[email protected]:5061>;tag=as65b9dc4d	
	275731	CSeq: 2 ACK	
	275732	User-Agent: Groundwire/5.9.68 (build 2064866; iOS 17.2.1; arm64-neon)	
	275733	Content-Length: 0
10

Are these 2 contexts loaded?

What does this output? asterisk -rx ‘dialplan show from-internal’

11

Thank you guys for all your help! I am flying out to Europe tomorrow, and I am so stressed …

(Why ‘DIGIUM_PHONE_USERS’?)

[ Context 'from-internal' created by 'DIGIUM_PHONE_USERS' ]
  'auto_hint_1102' => hint: SIP/1102,CustomPresence:1102            [DIGIUM_PHONE_USERS]
  'auto_hint_1103' => hint: SIP/1103,CustomPresence:1103            [DIGIUM_PHONE_USERS]
  'auto_hint_1104' => hint: SIP/1104,CustomPresence:1104            [DIGIUM_PHONE_USERS]
  'auto_hint_1105' => hint: SIP/1105,CustomPresence:1105            [DIGIUM_PHONE_USERS]
  'auto_hint_4601' => hint: SIP/4601,CustomPresence:4601            [DIGIUM_PHONE_USERS]
  'auto_hint_4602' => hint: SIP/4602,CustomPresence:4602            [DIGIUM_PHONE_USERS]
  'auto_hint_4603' => hint: SIP/4603,CustomPresence:4603            [DIGIUM_PHONE_USERS]
  'auto_hint_4605' => hint: SIP/4605,CustomPresence:4605            [DIGIUM_PHONE_USERS]
  'auto_hint_4606' => hint: SIP/4606,CustomPresence:4606            [DIGIUM_PHONE_USERS]
  'auto_hint_4608' => hint: SIP/4608,CustomPresence:4608            [DIGIUM_PHONE_USERS]
  'auto_hint_4611' => hint: SIP/4611,CustomPresence:4611            [DIGIUM_PHONE_USERS]
  'auto_hint_4612' => hint: SIP/4612,CustomPresence:4612            [DIGIUM_PHONE_USERS]
  'auto_hint_4613' => hint: SIP/4613,CustomPresence:4613            [DIGIUM_PHONE_USERS]
  'auto_hint_4614' => hint: SIP/4614,CustomPresence:4614            [DIGIUM_PHONE_USERS]
  'auto_hint_4621' => hint: SIP/4621,CustomPresence:4621            [DIGIUM_PHONE_USERS]
  'auto_hint_4622' => hint: SIP/4622,CustomPresence:4622            [DIGIUM_PHONE_USERS]
  'auto_hint_4623' => hint: SIP/4623,CustomPresence:4623            [DIGIUM_PHONE_USERS]
  'auto_hint_4700' => hint: SIP/4700,CustomPresence:4700            [DIGIUM_PHONE_USERS]
  'auto_hint_4701' => hint: SIP/4701,CustomPresence:4701            [DIGIUM_PHONE_USERS]
  'auto_hint_4702' => hint: SIP/4702,CustomPresence:4702            [DIGIUM_PHONE_USERS]
  'auto_hint_4703' => hint: SIP/4703,CustomPresence:4703            [DIGIUM_PHONE_USERS]
  'auto_hint_4704' => hint: SIP/4704,CustomPresence:4704            [DIGIUM_PHONE_USERS]
  'auto_hint_4705' => hint: SIP/4705,CustomPresence:4705            [DIGIUM_PHONE_USERS]
  'auto_hint_4790' => hint: SIP/4790,CustomPresence:4790            [DIGIUM_PHONE_USERS]
  'auto_hint_4791' => hint: SIP/4791,CustomPresence:4791            [DIGIUM_PHONE_USERS]
  'auto_hint_4792' => hint: PJSIP/4792,CustomPresence:4792          [DIGIUM_PHONE_USERS]
  'auto_hint_4795' => hint: SIP/4795,CustomPresence:4795            [DIGIUM_PHONE_USERS]
  'auto_hint_4798' => hint: SIP/4798,CustomPresence:4798            [DIGIUM_PHONE_USERS]
  'auto_hint_4799' => hint: SIP/4799,CustomPresence:4799            [DIGIUM_PHONE_USERS]
  'auto_hint_4801' => hint: SIP/4801,CustomPresence:4801            [DIGIUM_PHONE_USERS]
  'auto_hint_4802' => hint: SIP/4802,CustomPresence:4802            [DIGIUM_PHONE_USERS]
  'auto_hint_4803' => hint: SIP/4803,CustomPresence:4803            [DIGIUM_PHONE_USERS]
  'auto_hint_4804' => hint: SIP/4804,CustomPresence:4804            [DIGIUM_PHONE_USERS]

-= 33 extensions (33 priorities) in 1 context. =-
12

These look to be incomplete. They should include the extension that isn’t found, which, from your INVITEs, should be xxxxxxxxxx unless you are doing a GoTo to change the initial extension to be other than the request URI user.

Getting this for from-internal is strange, as I think FreePBX catches all numbers in that context.

13

This is a call coming in through SIPSTATION.

Again, I did not touch the PBX at all; it was running fine until a few hours ago. No idea what happened.

<--- SIP read from UDP:192.159.66.3:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.159.66.3;rport;branch=z9hG4bKHyp7B19NmcceN
Max-Forwards: 69
From: "UNKNOWN" <sip:[email protected]>;tag=6Xe74jNUHyDeD
To: <sip:[email protected]:5060>
Call-ID: f1fa146e-4603-123d-df81-549f3509619c
CSeq: 79412628 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: SIPStation 2.11.3
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-A-Leg-Call-ID: [email protected]
X-FS-Support: update_display,send_info
P-Asserted-Identity: "UNKNOWN" <sip:[email protected]>

v=0
o=Sonus_UAC 761736 933626 IN IP4 67.231.13.23
s=SIP Media Capabilities
c=IN IP4 67.231.13.23
t=0 0
m=audio 38114 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=ptime:20
<------------->
[2024-02-14 12:47:21] VERBOSE[12233] chan_sip.c: --- (19 headers 12 lines) ---
[2024-02-14 12:47:21] VERBOSE[12233] chan_sip.c: Sending to 192.159.66.3:5060 (NAT)
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Sending to 192.159.66.3:5060 (NAT)
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Using INVITE request as basis request - f1fa146e-4603-123d-df81-549f3509619c
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found peer 'fpbx-1-DtbGNyxE2XBg' for 'xxxxxxxxxxx' from 192.159.66.3:5060
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] netsock2.c: Using SIP RTP TOS bits 184
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] netsock2.c: Using SIP RTP CoS mark 5
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Got SDP version 933626 and unique parts [Sonus_UAC 761736 IN IP4 67.231.13.23]
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found RTP audio format 0
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found RTP audio format 18
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found RTP audio format 101
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found audio description format PCMU for ID 0
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found audio description format G729 for ID 18
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Found audio description format telephone-event for ID 101
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Peer audio RTP is at port 67.231.13.23:38114
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Looking for yyyyyyyyyy in from-pstn (domain 192.168.1.151)
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 192.159.66.3:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.159.66.3;branch=z9hG4bKHyp7B19NmcceN;received=192.159.66.3;rport=5060
From: "UNKNOWN" <sip:[email protected]>;tag=6Xe74jNUHyDeD
To: <sip:[email protected]:5060>;tag=as7c1a821a
Call-ID: f1fa146e-4603-123d-df81-549f3509619c
CSeq: 79412628 INVITE
Server: FPBX-15.0.37.4(16.30.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2024-02-14 12:47:21] NOTICE[12233][C-00000018] chan_sip.c: Call from 'DtbGNyxE2XBg' (192.159.66.3:5060) to extension 'yyyyyyyyyy' rejected because extension not found in context 'from-pstn'.
[2024-02-14 12:47:21] VERBOSE[12233][C-00000018] chan_sip.c: Scheduling destruction of SIP dialog 'f1fa146e-4603-123d-df81-549f3509619c' in 6400 ms (Method: INVITE)
[2024-02-14 12:47:21] VERBOSE[12233] chan_sip.c: 
<--- SIP read from UDP:192.159.66.3:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.159.66.3;rport;branch=z9hG4bKHyp7B19NmcceN
Max-Forwards: 69
From: "UNKNOWN" <sip:[email protected]>;tag=6Xe74jNUHyDeD
To: <sip:[email protected]:5060>;tag=as7c1a821a
Call-ID: f1fa146e-4603-123d-df81-549f3509619c
CSeq: 79412628 ACK
Content-Length: 0
14

I had shortened it. It is outgoing

chan_sip.c: Call from ‘4611’ (192.168.1.112:5060) to extension ‘xxxxxxxxxx’ rejected because extension not found in context ‘from-internal’.

and incoming

chan_sip.c: Call from ‘DtbGNyxE2XBg’ (192.159.66.3:5060) to extension ‘yyyyyyyyyy’ rejected because extension not found in context ‘from-pstn’.

xxxxxxxxxx is the external number I call.

yyyyyyyyyy is the did that is called.

15

As I said, I though FreePBX had a fallback to catch an invalid number in from-internal, but, in any case, can you show us the outgoing route that is supposed to match xxx…xxx.

16

The routes have not changed.

image
image1588×405 26.9 KB

17

Whilst I believe this is behaving as though the dialplan failed to load, your redaction means that there are possible unredacted values that wouldn’t match any of these patterns.

Also, I don’t think your last patter will ever match.

18

Thanks David.

Yes, I believe some files are not loading.

I did not make any changes, and all extensions and external numbers being unavailable and failing must be something catastrophic, not just a small issue in the routing plan or so.

19

So do an fwconsole reload and look for significant errors in the Asterisk log.

Or, if you want to take a pot shot at the problem, try restoring from the last “good” backup.

20

Thanks for all your help!

I had rebooted, reloaded, etc., multiple times, but it did not change anything.

Yesterday, while I was gone for a few hours, it suddenly started working again.

I have checked the log files; nobody was on the system. (Also, I am the only one who has access besides Sangoma.)