Hi everybody! Thnxs for reading in advance
I’m driving me crazy, I can’t find a solution for my problem. I’ve struggled long hours, so I decided to ask for help

At first, I’ll tell you THE problem, all working ok, but incoming calls end because of nat (I think) after 30 seconds.
Asterisk has 2 network boards, static public ip, extensions connected from outside the site. One of the boards is connected to a sip trunk, I’ve NAT enabled. Audio for both sides
In a SIP debug, I see “Retransmitting #10 (NAT) to 111.111.3.10:5060:” in INVITEs, but I think that is wrong the contact info (I see the public IP, instead the eth1’s IP), so I think it is the problem, and I don’t know how to solve it

Hope somebody can help me

Thnx again

I changed IPs, just for security

Asterisk 11.25.3

2 network boards (eth0 network/internet, eth1 sip trunk)

eth0 Link encap:Ethernet HWaddr 78:2B:CB:AE:0E:16
inet addr:192.168.10.223 Bcast:192.168.10.255 Mask:255.255.255.0
inet6 addr: xxxx::7a2b:cbff:feae:e16/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:98692 errors:0 dropped:0 overruns:0 frame:0
TX packets:85077 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:16984481 (16.1 MiB) TX bytes:73086281 (69.7 MiB)
Interrupt:20 Memory:e1c00000-e1c20000

eth1 Link encap:Ethernet HWaddr 7C:8B:CA:00:3D:7C
inet addr:111.111.64.149 Bcast:111.111.64.151 Mask:255.255.255.252
inet6 addr: xxxx::7e8b:caff:fe00:3d7c/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:29770 errors:0 dropped:0 overruns:0 frame:0
TX packets:27269 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:6193913 (5.9 MiB) TX bytes:6111981 (5.8 MiB)

Trunk name: Telec
host=111.111.3.10
type=peer
context=from-trunk
fromdomain=111.111.64.149
disallow=all
allow=alaw

Public IP 222.222.234.123

route -n

Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
111.111.3.10 111.111.64.150 255.255.255.255 UGH 0 0 0 eth1
111.111.64.148 0.0.0.0 255.255.255.252 U 0 0 0 eth1
192.168.10.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 eth1
169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 eth0
0.0.0.0 192.168.10.1 0.0.0.0 UG 0 0 0 eth0

sip.conf

nat=yes
ALLOW_SIP_ANON=no
externip=222.222.234.123
localnet=192.168.10.0/24
localnet=111.111.64.148/30

SIP debug, an external call from 1166667777 to 1122223333, ext 4466

[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – Called SIP/4466
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – Connected line update to SIP/Telec-0000000e prevented.
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – SIP/4466-0000000f is ringing
[2020-06-04 20:19:42] VERBOSE[3787][C-0000000e] app_dial.c: – Connected line update to SIP/Telec-0000000e prevented.
[2020-06-04 20:19:42] VERBOSE[3787][C-0000000e] app_dial.c: – SIP/4466-0000000f answered SIP/Telec-0000000e
[2020-06-04 20:19:46] VERBOSE[2409] chan_sip.c: Retransmitting #5 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:19:50] VERBOSE[2409] chan_sip.c: Retransmitting #6 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:19:54] VERBOSE[2409] chan_sip.c: Retransmitting #7 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:19:58] VERBOSE[2409] chan_sip.c: Retransmitting #8 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:20:00] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.252:5060 —>

<------------->
[2020-06-04 20:20:02] VERBOSE[2409] chan_sip.c: Retransmitting #9 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:20:03] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘0gQAAC8WAAACBAAALxYAAJa26xmPlKr8wixeofhzyRdMoaLor7v3oik715/n3IFF@111.111.3.10’ Method: OPTIONS
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:111.111.3.10:5060 —>
OPTIONS sip:metaswitch@111.111.64.149:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+2183e4ea3ff7dd836b11c1e3bf15c7731+sip+2+b00889c8
From: sip:metaswitch@111.111.3.10:5060;tag=111.111.3.10+2+6eb16302+c68d2e13
Content-Length: 0
Supported: resource-priority, siprec, 100rel
To: sip:metaswitch@111.111.64.149
Contact: sip:941235c91a33e3b43cf7d85de76db36e@111.111.3.10
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Max-Forwards: 69
Call-ID: 0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10
CSeq: 699873511 OPTIONS
Organization: Metaswitch Networks
Accept: application/sdp, application/dtmf-relay

<------------->
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: — (13 headers 0 lines) —
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Sending to 111.111.3.10:5060 (NAT)
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Looking for metaswitch in from-sip-external (domain 111.111.64.149)
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 111.111.3.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+2183e4ea3ff7dd836b11c1e3bf15c7731+sip+2+b00889c8;received=111.111.3.10;rport=5060
From: sip:metaswitch@111.111.3.10:5060;tag=111.111.3.10+2+6eb16302+c68d2e13
To: sip:metaswitch@111.111.64.149;tag=as371684c3
Call-ID: 0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10
CSeq: 699873511 OPTIONS
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:222.222.234.123:5060
Accept: application/sdp
Content-Length: 0

<------------>
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Scheduling destruction of SIP dialog ‘0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10’ in 32000 ms (Method: OPTIONS)
[2020-06-04 20:20:06] VERBOSE[2409] chan_sip.c: Retransmitting #10 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2020-06-04 20:20:07] WARNING[2409] chan_sip.c: Retransmission timeout reached on transmission 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10 for seqno 56524576 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[2020-06-04 20:20:07] WARNING[2409] chan_sip.c: Hanging up call 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [h@macro-dial-one:1] Macro(“SIP/Telec-0000000e”, “hangupcall,”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:1] ExecIf(“SIP/Telec-0000000e”, “0?Set(CDR(recordingfile)=.wav)”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:2] GotoIf(“SIP/Telec-0000000e”, “1?theend”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Goto (macro-hangupcall,s,4)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/Telec-0000000e”, “”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘hangupcall’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/Telec-0000000e’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Scheduling destruction of SIP dialog ‘2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060’ in 6400 ms (Method: INVITE)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: Parsing sip:4466@192.168.10.36:60798 for address/port to send to
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: set destination to 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.36:60798:
BYE sip:4466@192.168.10.36:60798 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK2a54c60e;rport
Max-Forwards: 70
From: “1166667777” sip:1166667777@192.168.10.223;tag=as6604fad6
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=aa39c34a
Call-ID: 2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060
CSeq: 103 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-dial-one, s, 45) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘dial-one’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘exten-vm’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: == Spawn extension (ext-local, 4466, 2) exited non-zero on ‘SIP/Telec-0000000e’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Scheduling destruction of SIP dialog ‘0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10’ in 32000 ms (Method: INVITE)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: Parsing sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10 for address/port to send to
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: set destination to 111.111.3.10:5060
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Reliably Transmitting (NAT) to 111.111.3.10:5060:
BYE sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10 SIP/2.0
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK42f5707d;rport
Max-Forwards: 70
From: sip:1122223333@111.111.64.149;tag=as2e963a0b
To: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK2a54c60e;rport=5060
Contact: sip:4466@192.168.10.36:60798
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=aa39c34a
From: “1166667777” sip:1166667777@192.168.10.223;tag=as6604fad6
Call-ID: 2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060
CSeq: 103 BYE
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (9 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060’ Method: INVITE
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:111.111.3.10:5060 —>
SIP/2.0 200 OK
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 102 BYE
From: sip:1122223333@111.111.64.149;tag=as2e963a0b
To: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
Via: SIP/2.0/UDP 222.222.234.123:5060;received=111.111.64.149;rport=5060;branch=z9hG4bK42f5707d
Content-Length: 0
Supported: resource-priority, siprec, 100rel
Contact: sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10
Server: DC-SIP/2.0
Organization: Metaswitch Networks
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (12 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409][C-0000000e] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10’ Method: INVITE
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
PUBLISH sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—ba8a6e7aed23642f
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=0b20e91e
Call-ID: y-qUP1725REoFNjQnRZg0Q…
CSeq: 1 PUBLISH
Expires: 600
Content-Type: application/pidf+xml
User-Agent: Z 3.15.40006 rv2.8.20
Event: presence
Allow-Events: presence, kpml, talk
Content-Length: 262

open Online

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (14 headers 3 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—ba8a6e7aed23642f;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=0b20e91e
To: sip:4466@192.168.10.223;transport=UDP;tag=as5edf2150
Call-ID: y-qUP1725REoFNjQnRZg0Q…
CSeq: 1 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘y-qUP1725REoFNjQnRZg0Q…’ Method: PUBLISH
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SUBSCRIBE sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—7acf366feff11173
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
User-Agent: Z 3.15.40006 rv2.8.20
Event: presence.winfo
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Creating new subscription
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: list_route: hop: sip:4466@192.168.10.36:60798;transport=UDP
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Found peer ‘4466’ for ‘4466’ from 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—7acf366feff11173;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
To: sip:4466@192.168.10.223;transport=UDP;tag=as7dc61397
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 1 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7516b3e8”
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Scheduling destruction of SIP dialog ‘QqCMBDM_dVMSM5fZ5HyN2A…’ in 6400 ms (Method: SUBSCRIBE)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SUBSCRIBE sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—d7a542af6bfdfbb6
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
User-Agent: Z 3.15.40006 rv2.8.20
Authorization: Digest username=“4466”,realm=“asterisk”,nonce=“7516b3e8”,uri="sip:4466@192.168.10.223;transport=UDP",response=“9aac98c51da5114be076284f7b0b3393”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (15 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Creating new subscription
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Found peer ‘4466’ for ‘4466’ from 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—d7a542af6bfdfbb6;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
To: sip:4466@192.168.10.223;transport=UDP;tag=as7dc61397
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘QqCMBDM_dVMSM5fZ5HyN2A…’ Method: SUBSCRIBE
[2020-06-04 20:20:08] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.132:5060 —>

<------------->
[2020-06-04 20:20:08] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘D4BFEBF0-3@111.111.3.10:5060’ Method: OPTIONS
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 24.232.134.32:55974:
OPTIONS sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK7f1ffec9;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@222.222.234.123;tag=as641529a1
To: sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP
Contact: sip:Unknown@222.222.234.123:5060
Call-ID: 229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:24.232.134.32:55974 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK7f1ffec9;rport=5060
Contact: sip:24.232.134.32:55974
To: sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP;tag=dc3f5348
From: "Unknown"sip:Unknown@222.222.234.123;tag=as641529a1
Call-ID: 229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060’ Method: OPTIONS
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.36:60798:
OPTIONS sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK04d8555b;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.10.223;tag=as1f1562f8
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP
Contact: sip:Unknown@192.168.10.223:5060
Call-ID: 4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK04d8555b;rport=5060
Contact: sip:192.168.10.36:60798
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=f45e8623
From: “Unknown” sip:Unknown@192.168.10.223;tag=as1f1562f8
Call-ID: 4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060’ Method: OPTIONS
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.252:5060:
OPTIONS sip:900@192.168.10.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK7c81c2d7;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.10.223;tag=as2a1cc252
To: sip:900@192.168.10.252:5060
Contact: sip:Unknown@192.168.10.223:5060
Call-ID: 78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.252:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK7c81c2d7;rport=5060
From: “Unknown” sip:Unknown@192.168.10.223;tag=as2a1cc252
To: sip:900@192.168.10.252:5060;tag=3991286321
Call-ID: 78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.130
Content-Length: 0

<------------->
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: — (8 headers 0 lines) —
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060’ Method: OPTIONS

Hope this helps Aaron…

Screenshot 2020-06-05 at 10.02.13.png2458×996 172 KB

Hi Everybody,

I’m testing Zulu for rolout on Customer site later.
I succeeded to install Zulu on FreePBX15 with Asterisk13 with certificate etc. all fine.
I also can use Zulu Client on Ubuntu 16, was a bit painfull as the latest Cliet is not working, but some releases back (3.2.1), is working fine now on Ubuntu 16.
So far I’m happy, all looking good and the Quality even from outside, seems to be good, as far I could Test.
I have tested a lot in the Past on Client and cellphone, Zoiper etc. and so far Zulu looks the best.
I also like, that Zulu is not working on any common SIP or IAX or WebRTC Port, but on 8002, whcih makes it more easy and save …
Last is the Android App, where I got a Problem I can’t solve.
I have an Oneplus 5 with Android9, not the newest, but also not to old …
I could enable VoLTE and Wifi Calling, but when I start Zulu, it pops up with an Messagewindow, “Action Required - Zulu must be enabled in your Android Phone App’s Calling Account Settings in orer to make and receive calls - cancel / Take me.”
If I press cancel, I’m logged in to teh Server, Chat works etc, but no calling in I press Take me, he jumps into me calling accounts, where I have 2 SIM Cards, one in German Telecom and one is China Telecom but from here i don’t know, how to proceed … no option to anable VoIP or similar …
As fare as I know, at least German Telecom has open VoLTE and Zoiper is working fine for me …
And, but at least on Wifi it should work …
Does anybody has an Idea ?
with many thanks in advance, Niels

That screen is the ‘simplified’ layout for PBXact users. PBXact is the paid commercial PBX product that extends the free FreePBX product. You can see the PBXact at the bottom of the screen cap.

Why is the agent timeout set to unlimited? This is probably the issue.
You’d want it to try other agents. Right?

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

Looking at the log, this seems to be a bug.

Please try to update your PBX, and if it’s up to date, please report this bug: issues.freepbx.org

Schedule a backup using the GUI and then look at the cron task generated with

crontab -l -uasterisk

I see now that the Wiki was updated and it’s now asking users to send an email VS submitting via the issue tracker.

Hi

from my testing and reports from customers its an issue with the email on outbound routes, Ive raised a Bug https://issues.freepbx.org/browse/FREEPBX-21565

and if you do the OPPOSITE to what is being said here but rever ‘core’ module back to 14.0.28.60 if on 14 ir what ever was core in 15 before teh change was made to add routes and all should work till bug is resolved.

heres the dodgy code and example of console out put

14.0.28.61: FREEI-1284: added feature to send out an email when an Outbound Route is dialed

Works as no email macro
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}b(func-apply-sipheaders^s^1))

Fails as email macro called
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}b(func-apply-sipheaders^s^1,(${DIAL_TRUNK}))M(send-obroute-email^${DIAL_NUMBER}^${MACRO_EXTEN}^${DIAL_TRUNK}^${NOW}^${CALLERID(name)}^${CALLERID(number)}))

-- IAX2/TRUNK_out-11079 is making progress passing it to Local/901234123456@from-queue-000000d2;2
-- IAX2/TRUNK_out-11079 is making progress passing it to Local/901234123456@from-queue-000000d2;2
-- IAX2/TRUNK_out-11079 is ringing
-- IAX2/TRUNK_out-11079 is making progress passing it to Local/901234123456@from-queue-000000d2;2
-- Local/901234123456@from-queue-000000d2;1 is ringing
-- IAX2/TRUNK_out-11079 answered Local/901234123456@from-queue-000000d2;2
-- Executing [s@macro-confirm:1] Set("IAX2/TRUNK_out-11079", "LOOPCOUNT=0") in new stack
-- Executing [s@macro-confirm:2] Set("IAX2/TRUNK_out-11079", "__MACRO_RESULT=ABORT") in new stack
-- Executing [s@macro-confirm:3] NoOp("IAX2/TRUNK_out-11079", "default and arv= ") in new stack
-- Executing [s@macro-confirm:4] ExecIf("IAX2/TRUNK_out-11079", "1?Set(ARG1=)") in new stack
-- Executing [s@macro-confirm:5] ExecIf("IAX2/TRUNK_out-11079", "1?Set(ALT_CONFIRM_MSG=)") in new stack
-- Executing [s@macro-confirm:6] Set("IAX2/TRUNK_out-11079", "MSG1=incoming-call-1-accept-2-decline") in new stack
-- Executing [s@macro-confirm:7] BackGround("IAX2/TRUNK_out-11079", "incoming-call-1-accept-2-decline,m,en,macro-confirm") in new stack
-- <IAX2/TRUNK_out-11079> Playing 'incoming-call-1-accept-2-decline.slin' (language 'en')



-- IAX2/TRUNK_out-82 is making progress passing it to Local/901234123456@from-queue-000000d3;2
-- IAX2/TRUNK_out-82 is making progress passing it to Local/901234123456@from-queue-000000d3;2
-- IAX2/TRUNK_out-82 is ringing
-- Local/901234123456@from-queue-000000d3;1 is ringing
-- IAX2/TRUNK_out-82 is making progress passing it to Local/901234123456@from-queue-000000d3;2
-- IAX2/TRUNK_out-82 answered Local/901234123456@from-queue-000000d3;2
-- Executing [s@macro-send-obroute-email:1] GotoIf("IAX2/TRUNK_out-82", "0?sendEmail") in new stack
-- Executing [s@macro-send-obroute-email:2] NoOp("IAX2/TRUNK_out-82", "email notifications disabled..exiting.") in new stack
-- Executing [s@macro-send-obroute-email:3] MacroExit("IAX2/TRUNK_out-82", "") in new stack
-- Local/901234123456@from-queue-000000d3;1 answered SIP/2000-00000074
-- Stopped music on hold on SIP/2000-00000074
-- Channel IAX2/TRUNK_out-82 joined 'simple_bridge' basic-bridge <2b2a3557-a8c7-4ab9-b671-98641b101a3a>
-- Channel Local/901234123456@from-queue-000000d3;1 joined 'simple_bridge' basic-bridge <d601f7bc-22ad-4bfa-b042-f8d1d7b6d2b8>
-- Channel Local/901234123456@from-queue-000000d3;2 joined 'simple_bridge' basic-bridge <2b2a3557-a8c7-4ab9-b671-98641b101a3a>
-- Channel SIP/2000-00000074 joined 'simple_bridge' basic-bridge <d601f7bc-22ad-4bfa-b042-f8d1d7b6d2b8>

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Despite having it set to unlimited it does try other agents which of course is what I need it to do.

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Thanks for all the help everyone. I have narrowed this down to an actual networking problem… Your guidance certainly helped. To Bill that suggested pjsip - this provider specifically doesn’t support pjsip, only legacy sip - so that wont help this time, but thanks for the reply. To the others, I think flipping the default gateway to the external interface will be the answer - but now that I have done that, I can see this connection isn’t going anywhere anyway. The traceroutes and pings were being rerouted over the internal LAN connection… I missed that part. Thanks all - ill post back when I finally get it.

Hi @chris_unit what about the codecs ?
Did you try with another Softphone, like Zoiper ?

Change it to how long you want it to ring by each agent, and see if it happens again