External sip peers unreachable

Hello. I am attempting to change ISPs. When I route traffic to and from Asterisk with the new ISP, it all works fine, for a while! After a few hours, all the external SIP peers become unreachable. I can also reproduce the problem by restarting Asterisk, or by reloading sip.conf. I can remedy the problem by turning off the external Internet connection for a few minutes, then turning it back on on again. The external sip peers become reachable again, and everything is fine for a few hours. IAX peers are not affected at all - they remain reachable and stable. I thought it might be a firewall problem, but posting to the Shorewall support list has not revealed any anomaly. I have tried with and without the Shorewall SIP helper - no difference.

Below is the SIP dialog when the peers are unreachable. To cut down on noise, I have set debugging on for just one peer.

[Jun 19 09:59:40] Asterisk 13.15.0 built by root @ pbx on a i686 running Linux on 2017-04-23 08:30:44 UTC
[Jun 19 09:59:48] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK7741821c;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as530fb99e
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.71b0
Call-ID: 64b523792366beb04d6400773c6a1320@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 09:59:48] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 09:59:48] VERBOSE[15434] chan_sip.c: Retransmitting #2 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK3075ad77
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 13:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 09:59:49] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK7741821c;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as530fb99e
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.71b0
Call-ID: 64b523792366beb04d6400773c6a1320@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 09:59:49] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 09:59:49] VERBOSE[15434] chan_sip.c: Retransmitting #3 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK3075ad77
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 13:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 09:59:50] VERBOSE[15434] chan_sip.c: Retransmitting #4 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK3075ad77
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 13:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 09:59:50] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK7741821c;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as530fb99e
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.71b0
Call-ID: 64b523792366beb04d6400773c6a1320@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 09:59:50] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 09:59:50] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK7741821c;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as530fb99e
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.71b0
Call-ID: 64b523792366beb04d6400773c6a1320@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 09:59:50] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 09:59:51] VERBOSE[15434] chan_sip.c: Retransmitting #5 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK3075ad77
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 13:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 09:59:52] VERBOSE[15434] chan_sip.c: Retransmitting #6 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK3075ad77
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 13:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 09:59:52] VERBOSE[15434] chan_sip.c: Really destroying SIP dialog '5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net' Method: OPTIONS
[Jun 19 09:59:55] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Beaune 52" <sip:52_89H7Fwf63e@asterisk.mydomain.net>;tag=9854333'
[Jun 19 09:59:56] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Beaune 52" <sip:52_89H7Fwf63e@asterisk.mydomain.net>;tag=9854333'
[Jun 19 10:00:00] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Beaune 52" <sip:52_89H7Fwf63e@asterisk.mydomain.net>;tag=9854333'
[Jun 19 10:00:01] NOTICE[15434] chan_sip.c: Outbound Registration: Expiry for slingshot.vtnoc.net is 300 sec (Scheduling reregistration in 285 s)
[Jun 19 10:00:02] VERBOSE[15434] chan_sip.c: Reliably Transmitting (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:03] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:03] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:03] VERBOSE[15434] chan_sip.c: Retransmitting #1 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:04] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:04] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:04] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Beaune 52" <sip:52_89H7Fwf63e@asterisk.mydomain.net>;tag=9854333'
[Jun 19 10:00:04] VERBOSE[15434] chan_sip.c: Retransmitting #2 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:05] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0

<------------->
[Jun 19 10:00:05] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:05] VERBOSE[15434] chan_sip.c: Retransmitting #3 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:06] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:06] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:06] VERBOSE[15434] chan_sip.c: Retransmitting #4 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:07] VERBOSE[15434] chan_sip.c: Retransmitting #5 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:07] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:07] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:08] VERBOSE[15434] chan_sip.c: Retransmitting #6 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK46a31aa4
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:08] VERBOSE[15434] chan_sip.c: Really destroying SIP dialog '244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net' Method: OPTIONS
[Jun 19 10:00:08] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:08] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:09] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK3075ad77;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as76e715db
To: <sip:sip.digiumcloud.net>;tag=576475fcb04af9dc1de68a3311691882.282c
Call-ID: 5a3353541c95e0cf0a0f367369225acf@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:09] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:10] NOTICE[15434] chan_sip.c: Still have a QUALIFY dialog active, deleting
[Jun 19 10:00:11] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 41" <sip:41#hsTy8d5s9R@asterisk.mydomain.net>;tag=e5321959a
879c5ffo0'
[Jun 19 10:00:12] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 40" <sip:40#hjdhezdh5Q@asterisk.mydomain.net>;tag=f2b16e04a
28101edo0'
[Jun 19 10:00:13] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 40" <sip:40#hjdhezdh5Q@asterisk.mydomain.net>;tag=f2b16e04a
28101edo0'
[Jun 19 10:00:13] NOTICE[15434] chan_sip.c: Peer '41#hsTy8d5s9R' is now UNREACHABLE!  Last qualify: 931
[Jun 19 10:00:16] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 41" <sip:41#hsTy8d5s9R@asterisk.mydomain.net>;tag=e5321959a
879c5ffo0'
[Jun 19 10:00:18] VERBOSE[15434] chan_sip.c: Reliably Transmitting (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK2e5fcd14
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as676b4bed
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 430034a206c2cf934aa8255e18b4e1ae@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:18] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 40" <sip:40#hjdhezdh5Q@asterisk.mydomain.net>;tag=f2b16e04a28101edo0'
[Jun 19 10:00:18] NOTICE[15434] chan_sip.c: Peer '53_frac8mewRu' is now UNREACHABLE!  Last qualify: 1407
[Jun 19 10:00:19] VERBOSE[15434] chan_sip.c:
<--- SIP read from UDP:8.17.32.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK46a31aa4;rport=5060
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as1d457251
To: <sip:sip.digiumcloud.net>;tag=cbbe90a9e3199009e192d1ff29429427.55c3
Call-ID: 244a98e421cfc2766495c82f064e7046@asterisk.mydomain.net
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: kamailio (4.1.3 (x86_64/linux))
Content-Length: 0


<------------->
[Jun 19 10:00:19] VERBOSE[15434] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 19 10:00:19] VERBOSE[15434] chan_sip.c: Retransmitting #1 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK2e5fcd14
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as676b4bed
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 430034a206c2cf934aa8255e18b4e1ae@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[Jun 19 10:00:20] NOTICE[15434] chan_sip.c: Correct auth, but based on stale nonce received from '"Chicago 41" <sip:41#hsTy8d5s9R@asterisk.mydomain.net>;tag=e5321959a879c5ffo0'
[Jun 19 10:00:20] VERBOSE[15434] chan_sip.c: Retransmitting #2 (no NAT) to 8.17.32.12:5060:
OPTIONS sip:sip.digiumcloud.net SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.99:5060;branch=z9hG4bK2e5fcd14
Max-Forwards: 70
From: "asterisk" <sip:58b9d0c80bf041b68c08473d93bc941c@asterisk.mydomain.net>;tag=as676b4bed
To: <sip:sip.digiumcloud.net>
Contact: <sip:58b9d0c80bf041b68c08473d93bc941c@xx.xx.xx.99:5060>
Call-ID: 430034a206c2cf934aa8255e18b4e1ae@asterisk.mydomain.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.0
Date: Mon, 19 Jun 2017 14:00:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---


Help appreicated!
Regards
Ian

Asterisk is sending OPTIONs, but the peer is not replying, or the request or replies are getting lost, in the network. Possibly an automatic NAT or firewall rule has timed out. There is no evidence of anything wrong with Asterisk.

Thanks for that analysis. I have tried running Asterisk on the router itself, and the peers remain reachable. So it does indeed seem to be a NAT issue.

Regards