The sip trace is not trimmed.I can’t find anything in the patton cfg related to the behaviour you just pointed out. Actually we were experimenting with fail2ban on this PBX recently, but I made sure to uninstall everything before these tests I’m making. It’s weird because the other patton does not have any problems whatsoever, even though it has a different interface (FXO vs ISDN)…
Anyways, I switched on the previous PBX we had until last month, with a different version of Asterisk that made use of chan_sip instead of res_pjsip, and everything works again, even the “problematic” ISDN patton. I can’t keep that old machine up though, I have to migrate the PBX to the new machine with res_pjsip.
I just wanted to make a quick test, this is the SIP trace of what happens in the old PBX with chan_sip (Patton’s IP is 172.16.1.235):
<--- SIP read from UDP:172.16.1.235:5060 --->
INVITE sip:01119703XXX@172.16.1.233 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.235:5060;branch=z9hG4bK510a13fb59aa132d1
Max-Forwards: 70
From: <sip:3357897XXX@172.16.1.235:5060>;tag=4d00fb63e2
To: <sip:01119703XXX@172.16.1.233>
Call-ID: 5c25d9d83564e7d9
CSeq: 4086 INVITE
Contact: <sip:3357897XXX@172.16.1.235:5060;transport=udp>
P-Preferred-Identity: <sip:3357897XXX@172.16.1.235:5060>
Supported: replaces
User-Agent: Patton SN4120 1BIS2V 00A0BA0A8DB4 R6.3 2013-05-01 H323 SIP M5T SIP Stack/4.1.12.18
Content-Type: application/sdp
Content-Length: 265
v=0
o=MxSIP 0 6 IN IP4 172.16.1.235
s=SIP Call
c=IN IP4 172.16.1.235
t=0 0
m=audio 4868 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Sending to 172.16.1.235:5060 (NAT)
Sending to 172.16.1.235:5060 (NAT)
Using INVITE request as basis request - 5c25d9d83564e7d9
Found peer '10001' for '3357897XXX' from 172.16.1.235:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f5fb8017e00 -- Strict RTP learning after remote address set to: 172.16.1.235:4868
Peer audio RTP is at port 172.16.1.235:4868
Looking for 01119703XXX in incoming (domain 172.16.1.233)
sip_route_dump: route/path hop: <sip:3357897XXX@172.16.1.235:5060;transport=udp>
<--- Transmitting (NAT) to 172.16.1.235:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.235:5060;branch=z9hG4bK510a13fb59aa132d1;received=172.16.1.235;rport=5060
From: <sip:3357897XXX@172.16.1.235:5060>;tag=4d00fb63e2
To: <sip:01119703XXX@172.16.1.233>
Call-ID: 5c25d9d83564e7d9
CSeq: 4086 INVITE
Server: Asterisk PBX 13.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01119703XXX@172.16.1.233:5060>
Content-Length: 0
<------------>
-- Executing [01119703XXX@incoming:1] Set("SIP/10001-0000000a", "FROM_DID=01119703XXX") in new stack
-- Executing [01119703XXX@incoming:2] Set("SIP/10001-0000000a", "CALLERID(name)=MA 3357897XXX") in new stack
-- Executing [01119703XXX@incoming:3] Dial("SIP/10001-0000000a", "SIP/801,60,tTr") in new stack
== Using SIP RTP CoS mark 5
Audio is at 15722
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.16.1.234:5060:
INVITE sip:801@172.16.1.234:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.233:5060;branch=z9hG4bK3b8cc2db;rport
Max-Forwards: 70
From: "MA 3357897XXX" <sip:3357897XXX@172.16.1.233>;tag=as6417c866
To: <sip:801@172.16.1.234:5060>
Contact: <sip:3357897XXX@172.16.1.233:5060>
Call-ID: 39819aa16278ccb827ccc8811c91446b@172.16.1.233:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.17.2
Date: Tue, 14 Jun 2022 12:44:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 274
v=0
o=root 137388565 137388565 IN IP4 172.16.1.233
s=Asterisk PBX 13.17.2
c=IN IP4 172.16.1.233
t=0 0
m=audio 15722 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/801
<--- Transmitting (NAT) to 172.16.1.235:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.1.235:5060;branch=z9hG4bK510a13fb59aa132d1;received=172.16.1.235;rport=5060
From: <sip:3357897XXX@172.16.1.235:5060>;tag=4d00fb63e2
To: <sip:01119703XXX@172.16.1.233>;tag=as0be3785c
Call-ID: 5c25d9d83564e7d9
CSeq: 4086 INVITE
Server: Asterisk PBX 13.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01119703XXX@172.16.1.233:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:172.16.1.234:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.233:5060;branch=z9hG4bK3b8cc2db;rport=5060
From: "MA 3357897XXX" <sip:3357897XXX@172.16.1.233>;tag=as6417c866
To: <sip:801@172.16.1.234:5060>
Call-ID: 39819aa16278ccb827ccc8811c91446b@172.16.1.233:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.7
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:172.16.1.234:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.1.233:5060;branch=z9hG4bK3b8cc2db;rport=5060
From: "MA 3357897XXX" <sip:3357897XXX@172.16.1.233>;tag=as6417c866
To: <sip:801@172.16.1.234:5060>;tag=1348042945
Call-ID: 39819aa16278ccb827ccc8811c91446b@172.16.1.233:5060
CSeq: 102 INVITE
Contact: <sip:801@172.16.1.234:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.3.7
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:801@172.16.1.234:5060>
-- SIP/801-0000000b is ringing
<--- Transmitting (NAT) to 172.16.1.235:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.1.235:5060;branch=z9hG4bK510a13fb59aa132d1;received=172.16.1.235;rport=5060
From: <sip:3357897XXX@172.16.1.235:5060>;tag=4d00fb63e2
To: <sip:01119703XXX@172.16.1.233>;tag=as0be3785c
Call-ID: 5c25d9d83564e7d9
CSeq: 4086 INVITE
Server: Asterisk PBX 13.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01119703XXX@172.16.1.233:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:172.16.0.109:5060 --->
REGISTER sip:MA SIP/2.0
Via: SIP/2.0/UDP 172.16.0.109:5060;branch=z9hG4bKa11c4f4fc5628cc539d49ddba1e755d6;rport
From: "202" <sip:202@MA>;tag=1167122267
To: "202" <sip:202@MA>
Call-ID: 3129382509@172_16_0_109
CSeq: 321 REGISTER
Contact: <sip:202@172.16.0.109:5060>
Max-Forwards: 70
User-Agent: C610 IP/42.243.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 172.16.0.109:5060 (NAT)
Sending to 172.16.0.109:5060 (NAT)
<--- Transmitting (NAT) to 172.16.0.109:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.109:5060;branch=z9hG4bKa11c4f4fc5628cc539d49ddba1e755d6;received=172.16.0.109;rport=5060
From: "202" <sip:202@MA>;tag=1167122267
To: "202" <sip:202@MA>;tag=as47fccc35
Call-ID: 3129382509@172_16_0_109
CSeq: 321 REGISTER
Server: Asterisk PBX 13.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31b51db9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3129382509@172_16_0_109' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:172.16.0.109:5060 --->
REGISTER sip:MA SIP/2.0
Via: SIP/2.0/UDP 172.16.0.109:5060;branch=z9hG4bKf5e99562eef17486c354e84b76c5f5e9;rport
From: "202" <sip:202@MA>;tag=1167122267
To: "202" <sip:202@MA>
Call-ID: 3129382509@172_16_0_109
CSeq: 322 REGISTER
Contact: <sip:202@172.16.0.109:5060>
Authorization: Digest username="202", realm="asterisk", algorithm=MD5, uri="sip:MA", nonce="31b51db9", response="7e5edf7d3f778a5d086182980f3dc444"
Max-Forwards: 70
User-Agent: C610 IP/42.243.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.0.109:5060 (NAT)
Reliably Transmitting (NAT) to 172.16.0.109:5060:
OPTIONS sip:202@172.16.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.233:5060;branch=z9hG4bK283bfacb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.1.233>;tag=as70828cd0
To: <sip:202@172.16.0.109:5060>
Contact: <sip:asterisk@172.16.1.233:5060>
Call-ID: 3c6fe8d867c1aa6f41f7126522906a83@172.16.1.233:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.17.2
Date: Tue, 14 Jun 2022 12:44:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 172.16.0.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.109:5060;branch=z9hG4bKf5e99562eef17486c354e84b76c5f5e9;received=172.16.0.109;rport=5060
From: "202" <sip:202@MA>;tag=1167122267
To: "202" <sip:202@MA>;tag=as47fccc35
Call-ID: 3129382509@172_16_0_109
CSeq: 322 REGISTER
Server: Asterisk PBX 13.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 180
Contact: <sip:202@172.16.0.109:5060>;expires=180
Date: Tue, 14 Jun 2022 12:44:12 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3129382509@172_16_0_109' in 32000 ms (Method: REGISTER)
Retransmitting #4 (NAT) to 94.32.130.112:5060:
OPTIONS sip:ims.tiscali.net SIP/2.0
Via: SIP/2.0/UDP 188.218.155.110:5060;branch=z9hG4bK4552c32c;rport
Max-Forwards: 70
From: "asterisk" <sip:00390115849478@188.218.155.110>;tag=as356e31c2
To: <sip:ims.tiscali.net>
Contact: <sip:00390115849478@188.218.155.110:5060>
Call-ID: 1d5d184f269d659a471aa3340da481b8@188.218.155.110:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.17.2
Date: Tue, 14 Jun 2022 12:44:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '1d5d184f269d659a471aa3340da481b8@188.218.155.110:5060' Method: OPTIONS
<--- SIP read from UDP:172.16.0.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.233:5060;branch=z9hG4bK283bfacb;rport=5060
From: "asterisk" <sip:asterisk@172.16.1.233>;tag=as70828cd0
To: <sip:202@172.16.0.109:5060>;tag=ar61939be1
Call-ID: 3c6fe8d867c1aa6f41f7126522906a83@172.16.1.233:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: C610 IP/42.243.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Accept: application/sdp, application/dtmf-relay, message/sipfrag, application/simple-message-summary, application/url, multipart/mixed
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '3c6fe8d867c1aa6f41f7126522906a83@172.16.1.233:5060' Method: OPTIONS
Really destroying SIP dialog 'a4dfb8819c52d0475f29c608b321d3fc@0:0:0:0:0:0:0:0' Method: OPTIONS
== Manager 'fop2' logged on from 127.0.0.1
== Manager 'fop2' logged off from 127.0.0.1
voiptop1*CLI> sip set debug off
SIP Debugging Disabled
== Spawn extension (incoming, 01119703XXX, 3) exited non-zero on 'SIP/10001-0000000a'