Responding to SIP messages on a different IP

I have an Asterisk 14.7 system that is communicating with an external PBX system. Our Asterisk system fails to respond to BYE requests sent by this external PBX. I believe it is because this external system is using different IPs for inbound and outbound SIP traffic. With ‘sip set debug on’, our Asterisk system fails to see the BYE request; I have to use tcpdump to see the requests coming in. I can see the BYE request Call-ID match up to a channel on Asterisk.

Is this functionality supported in Asterisk? How does Asterisk match BYE requests to channels?

Which SIP channel driver (SIP or PJSIP)?

Asterisk will reply to BYE even if it doesn’t match, although the reply may be a 4xx or 5xx response. The only reason it wouldn’t reply is if the reply was sent to an address/port combination that wasn’t bound to Asterisk.

You need to provide the actual logs with the absolute minimum redaction of addressses, and an explanation of how the redaction was done.

We are using SIP. I setup the SIP peer with the name of ‘test1’, and they gave us the number ‘1234’ to test. Attached is the tcpdump and Asterisk console log of the full call. It looks like they aren’t using their outbound external IP anymore so disregard that. I changed the following things:

  • Ext System IP to 30.1.1.1
  • Our External IP to 50.3.3.3
  • CallerID to dbrown
  • Caller Number to 2189791212
  • Hostname to asterisk.our-hostname.com

[Jun 12 21:43:54]   == Using SIP RTP CoS mark 5
[Jun 12 21:43:54]        > 0x7fc968018810 -- Strict RTP learning after remote address set to: 10.100.36.201:19574
[Jun 12 21:43:54]     -- Executing [EXT1234@inbound:1] Dial("SIP/10.100.36.201-00000002", "SIP/test1/1234") in new stack
[Jun 12 21:43:54]   == Using SIP RTP CoS mark 5
[Jun 12 21:43:54]     -- Called SIP/test1/1234
[Jun 12 21:43:54]        > 0x560c42fcb320 -- Strict RTP learning after remote address set to: 30.1.1.1:36166
[Jun 12 21:43:54]     -- SIP/test1-00000003 is making progress passing it to SIP/10.100.36.201-00000002
[Jun 12 21:43:54]        > 0x7fc968018810 -- Strict RTP switching to RTP target address 10.100.36.201:19574 as source
[Jun 12 21:43:54]        > 0x560c42fcb320 -- Strict RTP switching to RTP target address 30.1.1.1:36166 as source
[Jun 12 21:43:55]     -- SIP/test1-00000003 is ringing
[Jun 12 21:43:55]     -- SIP/test1-00000003 is making progress passing it to SIP/10.100.36.201-00000002
[Jun 12 21:43:55]     -- SIP/test1-00000003 answered SIP/10.100.36.201-00000002
[Jun 12 21:43:55]     -- Channel SIP/test1-00000003 joined 'simple_bridge' basic-bridge <1912f34f-94f8-41a0-9b90-2261893df41a>
[Jun 12 21:43:55]     -- Channel SIP/10.100.36.201-00000002 joined 'simple_bridge' basic-bridge <1912f34f-94f8-41a0-9b90-2261893df41a>
[Jun 12 21:43:55]        > 0x7fc968018810 -- Strict RTP learning complete - Locking on source address 10.100.36.201:19574
[Jun 12 21:43:55]        > 0x560c42fcb320 -- Strict RTP learning complete - Locking on source address 30.1.1.1:36166
asterisk-01*CLI> core show channel SIP/test1-00000003
 -- General --
           Name: SIP/test1-00000003
           Type: SIP
       UniqueID: asterisk-01-1591998234.4
       LinkedID: asterisk-01-1591998234.3
      Caller ID: 1234
 Caller ID Name: (N/A)
Connected Line ID: +2189791212
Connected Line ID Name: dbrown
Eff. Connected Line ID: +2189791212
Eff. Connected Line ID Name: dbrown
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
  NativeFormats: (ulaw)
    WriteFormat: ulaw
     ReadFormat: ulaw
 WriteTranscode: No
  ReadTranscode: No
 Time to Hangup: 0
   Elapsed Time: 0h0m31s
      Bridge ID: 1912f34f-94f8-41a0-9b90-2261893df41a
 --   PBX   --
        Context: inbound
      Extension:
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: AppDial
           Data: (Outgoing Line)
 Call Identifer: [C-00000002]
      Variables:
BRIDGEPVTCALLID=330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
BRIDGEPEER=SIP/10.100.36.201-00000002
DIALEDPEERNUMBER=test1/1234
SIPCALLID=262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
  CDR Variables:
level 1: calledsubaddr=
level 1: callingsubaddr=
level 1: dnid=
level 1: clid="" <1234>
level 1: src=1234
level 1: dcontext=inbound
level 1: channel=SIP/test1-00000003
level 1: lastapp=AppDial
level 1: lastdata=(Outgoing Line)
level 1: start=1591998234.145717
level 1: answer=1591998235.256629
level 1: end=1591998235.257487
level 1: duration=1
level 1: billsec=0
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=asterisk-01-1591998234.4
level 1: linkedid=asterisk-01-1591998234.3
level 1: sequence=3

[Jun 12 21:44:28]     -- Channel SIP/10.100.36.201-00000002 left 'simple_bridge' basic-bridge <1912f34f-94f8-41a0-9b90-2261893df41a>
[Jun 12 21:44:28]     -- Channel SIP/test1-00000003 left 'simple_bridge' basic-bridge <1912f34f-94f8-41a0-9b90-2261893df41a>
[Jun 12 21:44:28]   == Spawn extension (inbound, EXT1234, 1) exited non-zero on 'SIP/10.100.36.201-00000002'
21:43:54.139247 IP 10.100.36.201.5060 > 10.100.14.116.5060: SIP: INVITE sip:EXT1234@asterisk.our-hostname.com SIP/2.0
INVITE sip:EXT1234@asterisk.our-hostname.com SIP/2.0
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK1c5bcbd4
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>
Contact: <sip:+12189791212@10.100.36.201:5060>
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 102 INVITE
Date: Fri, 12 Jun 2020 21:43:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
callTag: 1294FC5C75D3B57D891AA38CC7486F04
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 40932760 40932760 IN IP4 10.100.36.201
s=Asterisk PBX 11.18.0
c=IN IP4 10.100.36.201
t=0 0
m=audio 19574 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

21:43:54.144457 IP 10.100.14.116.5060 > 10.100.36.201.5060: SIP: SIP/2.0 100 Trying
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK1c5bcbd4;received=10.100.36.201;rport=5060
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:EXT1234@10.100.14.116:5060>
Content-Length: 0


21:43:54.146097 IP 10.100.14.116.5060 > 30.1.1.1.5060: SIP: INVITE sip:1234@30.1.1.1 SIP/2.0
INVITE sip:1234@30.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK0c6b9e21
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>
Contact: <sip:+12189791212@50.3.3.3:5060>
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
CSeq: 102 INVITE
Date: Fri, 12 Jun 2020 21:43:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1330333786 1330333786 IN IP4 50.3.3.3
s=Asterisk PBX 14.7.8
c=IN IP4 50.3.3.3
t=0 0
m=audio 14462 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

21:43:54.222058 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: SIP/2.0 100 Trying
SIP/2.0 100 Trying
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>
CSeq: 102 INVITE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK0c6b9e21
Content-Length: 0


21:43:54.239828 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: SIP/2.0 183 Session Progress
SIP/2.0 183 Session Progress
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
CSeq: 102 INVITE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Contact: <sip:01234@30.1.1.1:5060>
Record-Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH, UPDATE
Supported: histinfo, join, replaces, timer
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK0c6b9e21
P-Asserted-Identity: <sip:01234@30.1.1.1>
Content-Type: application/sdp
Content-Length: 177

v=0
o=- 1591998261 2 IN IP4 30.1.1.1
s=-
c=IN IP4 30.1.1.1
b=AS:64
t=0 0
m=audio 36166 RTP/AVP 0 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=ptime:20

21:43:54.240563 IP 10.100.14.116.5060 > 10.100.36.201.5060: SIP: SIP/2.0 183 Session Progress
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK1c5bcbd4;received=10.100.36.201;rport=5060
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>;tag=as003df84f
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:EXT1234@10.100.14.116:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 239

v=0
o=root 1202316010 1202316010 IN IP4 10.100.14.116
s=Asterisk PBX 14.7.8
c=IN IP4 10.100.14.116
t=0 0
m=audio 19632 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

21:43:55.254805 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: SIP/2.0 180 Ringing
SIP/2.0 180 Ringing
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
CSeq: 102 INVITE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Contact: <sip:30.1.1.1:5060>
Record-Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH, UPDATE
Supported: histinfo, join, replaces, timer
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK0c6b9e21
P-Asserted-Identity: <sip:1234@30.1.1.1>
Content-Type: application/sdp
Content-Length: 177

v=0
o=- 1591998261 2 IN IP4 30.1.1.1
s=-
c=IN IP4 30.1.1.1
b=AS:64
t=0 0
m=audio 36166 RTP/AVP 0 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=ptime:20

21:43:55.256219 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: SIP/2.0 200 OK
SIP/2.0 200 OK
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
CSeq: 102 INVITE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Contact: <sip:30.1.1.1:5060>
Record-Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH, UPDATE
Supported: histinfo, join, replaces, timer
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK0c6b9e21
Require: timer
P-Asserted-Identity: <sip:1234@30.1.1.1>
Session-Expires: 1200;refresher=uas
Content-Type: application/sdp
Content-Length: 177

v=0
o=- 1591998261 2 IN IP4 30.1.1.1
s=-
c=IN IP4 30.1.1.1
b=AS:64
t=0 0
m=audio 36166 RTP/AVP 0 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=ptime:20

21:43:55.256516 IP 10.100.14.116.5060 > 30.1.1.1.5060: SIP: ACK sip:30.1.1.1:5060 SIP/2.0
ACK sip:30.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK39f01908
Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
Contact: <sip:+12189791212@50.3.3.3:5060>
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
CSeq: 102 ACK
Content-Length: 0


21:43:55.256927 IP 10.100.14.116.5060 > 10.100.36.201.5060: SIP: SIP/2.0 200 OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK1c5bcbd4;received=10.100.36.201;rport=5060
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>;tag=as003df84f
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:EXT1234@10.100.14.116:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 239

v=0
o=root 1202316010 1202316010 IN IP4 10.100.14.116
s=Asterisk PBX 14.7.8
c=IN IP4 10.100.14.116
t=0 0
m=audio 19632 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

21:43:55.257927 IP 10.100.36.201.5060 > 10.100.14.116.5060: SIP: ACK sip:EXT1234@10.100.14.116:5060 SIP/2.0
ACK sip:EXT1234@10.100.14.116:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK6b809057
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>;tag=as003df84f
Contact: <sip:+12189791212@10.100.36.201:5060>
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 102 ACK
Content-Length: 0


21:44:03.636664 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:04.137360 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:05.138114 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:07.138952 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:11.139566 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:15.140483 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:19.141170 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:23.141945 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:27.142905 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:28.440582 IP 10.100.36.201.5060 > 10.100.14.116.5060: SIP: BYE sip:EXT1234@10.100.14.116:5060 SIP/2.0
BYE sip:EXT1234@10.100.14.116:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK46abab5e
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>;tag=as003df84f
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 103 BYE
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


21:44:28.441010 IP 10.100.14.116.5060 > 10.100.36.201.5060: SIP: SIP/2.0 200 OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.36.201:5060;branch=z9hG4bK46abab5e;received=10.100.36.201;rport=5060
From: "dbrown" <sip:+12189791212@10.100.36.201>;tag=as37e9d95e
To: <sip:EXT1234@asterisk.our-hostname.com>;tag=as003df84f
Call-ID: 330e6e64601f8ce75d67b4107540d36f@10.100.36.201:5060
CSeq: 103 BYE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


21:44:28.442227 IP 10.100.14.116.5060 > 30.1.1.1.5060: SIP: BYE sip:30.1.1.1:5060 SIP/2.0
BYE sip:30.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK340bb1c6
Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Max-Forwards: 70
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
CSeq: 103 BYE
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


21:44:28.520560 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: SIP/2.0 200 OK
SIP/2.0 200 OK
From: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
To: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
CSeq: 103 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Record-Route: <sip:30.1.1.1:5060;ipcs-line=1655080;lr;transport=udp>
Supported: histinfo, join, replaces, timer
Via: SIP/2.0/UDP 50.3.3.3:5060;branch=z9hG4bK340bb1c6
Content-Length: 0


21:44:31.143649 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


21:44:35.144781 IP 30.1.1.1.5060 > 10.100.14.116.5060: SIP: BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
BYE sip:+12189791212@50.3.3.3:5060 SIP/2.0
From: <sip:1234@30.1.1.1>;tag=dfdd356acf541ea90ca0c29b0b1e3
To: "dbrown" <sip:+12189791212@50.3.3.3>;tag=as1d267676
CSeq: 1 BYE
Call-ID: 262c32b02775239d1d7096b654743d5d@50.3.3.3:5060
Supported: histinfo, join, replaces, timer
Max-Forwards: 66
Via: SIP/2.0/UDP 30.1.1.1:5060;branch=z9hG4bK-s1632-000464547203-1--s1632-
Content-Length: 0


What does the Asterisk SIP debugging show was received?

Could I also ask why you are not using PJSIP.

Where was tcpdump run?

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.