SIP Trunk problem - NAT - 2 networks

siptrunk
asterisk
Tags: #<Tag:0x00007f702e1a0b58> #<Tag:0x00007f702d6e3f80>

(Marco) #1

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

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:4466@192.168.10.223;transport=UDP"> <tuple id="4466" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
[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

#2

In Asterisk SIP Settings, try setting Local Networks to:
192.168.10.0 / 24
111.111.64.148 / 30
111.111.3.10 / 32

After changing these, restart (not just reload) Asterisk and test. If no luck, post a new SIP trace.

Doing this kind of stuff with a relatively ancient Asterisk without pjsip is tough, because you can’t set up different transports for the networks.


(Marco) #3

Yesssssssssssss!
Thanks a lot @Stewart1


(system) closed #4

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