SIP one way audio and call disconnecting

Asterisk Asterisk Support
1

I know this topic has been answered a lot of times but I have been researching for weeks and doing trial and error but my problem still exists.
I have Asterisk 11.5.1 running on Ubuntu Server 12.04 LTS. Locally, I can make calls in and out with my Google Voice account. My goal is to have another phone outside of the network to make and receive call using my asterisk server. For some reason, I have not able to make it work.
I have a Cisco 7940 outside of my network and it can receive and make calls but the issues that I am facing are one way audio and call disconnecting. From the cisco phone outside of the network, I can hear the person from the other line, they just cannot hear me.
I feel like I have tried everything but I know I have not and I followed multiple sources and still failed to make it work.
Basically, the server is behind a router and the cisco phone is behind a router as well so I have forwarded my RTP from 10000-32000, 5222 XMPP and 5060-5082 for SIP for both internal network (server) and external network (for the phone). I have tried setting up STUN along with it as well. Tried DMZ for the server. Tried turning off my firewall for my server.

Here are my configurations

[quote]rtp.conf
rtpstart=10000
rtpend=12000
icesupport=yes ; For google voice
[/quote]

sip.conf

This is me trying to call out
This will happen at least 6 tries and then next quote underneath will happen

[quote]<— Reliably Transmitting (NAT) to externalIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP
externalIP:5060;branch=z9hG4bK31c37e34;received=ExternalIP;rport=5060^M
From: “201” sip:201@insertdnshere.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:XXXXXX@insertdnshere.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:XXXXXXX@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M

v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M
[/quote]

Here’s what it says when it disconnects

[quote]Retransmission timeout reached on transmission 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14 for seqno 102 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions

Hanging up call 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).

[0K == Spawn extension (gvoice-makkugasho, XXXXXXX, 1) exited non-zero on ‘SIP/201-00000005’
Scheduling destruction of SIP dialog ‘000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14’ in 6400 ms (Method: INVITE)
[/quote]

After that a BYE packet will be sent…
Any ideas? I will try anything at this point. I am just desperate on making this work.
If you need my full sip debug on a call, please let me know.

2

What is the critical packet that it is retransmitting?

3

I believe the critical packet it was retransmitting was the log above that I posted. It transmits it at least 7 times.

Here is the full SIP debug when placing a call until it drops.

[quote]<— SIP read from UDP:EXTERNALIP:5060 —>
INVITE sip:3581619@mydns.com SIP/2.0
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK5b703d18
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae
To: sip:3581619@mydns.com
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:201@EXTERNALIP:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
<------------>
Scheduling destruction of SIP dialog ‘000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14’ in 6400 ms (Method: INVITE)
^M^[[K2850*CLI> ^M^[[0KRetransmitting #1 (NAT) to EXTERNALIP:5060:
SIP/2.0 401 Unauthorized^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK5b703d18;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as5836c58d^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 101 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“2cced021”^M
Content-Length: 0^M

^M^[[K2850*CLI> ^M^[[0K
<— SIP read from UDP:EXTERNALIP:5060 —>
ACK sip:3581619@mydns.com SIP/2.0
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK5b703d18
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae
To: sip:3581619@mydns.com;tag=as5836c58d
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
CSeq: 101 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
^M^[[K2850*CLI> ^M^[[0K
<— SIP read from UDP:EXTERNALIP:5060 —>
INVITE sip:3581619@mydns.com SIP/2.0
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae
To: sip:3581619@mydns.com
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:201@EXTERNALIP:5060;transport=udp
Authorization: Digest username=“201”,realm=“asterisk”,uri="sip:3581619@mydns.com",response=“cc69ef49c2186a903dc66027cae67509”,nonce=“2cced021”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “201” sip:201@mydns.com;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 222
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 14927 0 IN IP4 EXTERNALIP
s=SIP Call
t=0 0
m=audio 21812 RTP/AVP 18 0 8
c=IN IP4 EXTERNALIP
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
<------------->
— (18 headers 11 lines) —
^M^[[K2850*CLI> ^M^[[0KSending to EXTERNALIP:5060 (NAT)
Using INVITE request as basis request - 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
Found peer ‘201’ for ‘201’ from EXTERNALIP:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port EXTERNALIP:21812
Looking for 3581619 in gvoice-makkugasho (domain mydns.com)
list_route: hop: sip:201@EXTERNALIP:5060;transport=udp

<— Transmitting (NAT) to EXTERNALIP:5060 —>
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Length: 0^M
<------------>
– Executing [3581619@gvoice-makkugasho:1] ^[[1;36mDial^[[0m(“^[[1;35mSIP/201-00000005^[[0m”, “^[[1;35mMotif/gvoice-makkugasho/5203581619@voice.google.com,r^[[0m”) in new stack
^M^[[K2850*CLI> ^M^[[0K – Called Motif/gvoice-makkugasho/5203581619@voice.google.com

<— Transmitting (NAT) to EXTERNALIP:5060 —>
SIP/2.0 180 Ringing^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Length: 0^M
^M

<------------>
^M^[[K2850CLI> ^M^[[0K – Motif/5203581619@voice.google.com-a281 is proceeding passing it to SIP/201-00000005
^M^[[K2850CLI> ^M^[[0K > 0x7f4bd4006520 – Probation passed - setting RTP source address to 173.194.79.127:19305
^M^[[K2850CLI> ^M^[[0K – Motif/5203581619@voice.google.com-a281 answered SIP/201-00000005
^M^[[K2850CLI> ^M^[[0KAudio is at 10182
^M^[[K2850*CLI> ^M^[[0KAdding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP

<— Reliably Transmitting (NAT) to EXTERNALIP:5060 —>
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

<------------>
^M^[[K2850*CLI> ^M^[[0KRetransmitting #1 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #2 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #3 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #4 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #5 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #6 (NAT) to EXTERNALIP:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP EXTERNALIP:5060;branch=z9hG4bK31c37e34;received=EXTERNALIP;rport=5060^M
From: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
To: sip:3581619@mydns.com;tag=as1eb0402a^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: sip:3581619@192.168.1.100:5060^M
Content-Type: application/sdp^M
Content-Length: 205^M
^M
v=0^M
o=root 1071029617 1071029617 IN IP4 192.168.1.100^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 192.168.1.100^M
t=0 0^M
m=audio 10182 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=ptime:20^M
a=sendrecv^M

^M^[[K2850CLI> ^M^[[0K[Nov 26 22:39:09] ^[[1;31mWARNING^[[0m[4453]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m4174^[[0m ^[[1;37mretrans_pkt^[[0m: Retransmission timeout reached on transmission 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14 for seqno 102 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6399ms with no response
[Nov 26 22:39:09] ^[[1;31mWARNING^[[0m[4453]: ^[[1;37mchan_sip.c^[[0m:^[[1;37m4203^[[0m ^[[1;37mretrans_pkt^[[0m: Hanging up call 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
^M^[[K2850CLI> ^M^[[0K == Spawn extension (gvoice-makkugasho, 3581619, 1) exited non-zero on ‘SIP/201-00000005’
Scheduling destruction of SIP dialog ‘000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14’ in 6400 ms (Method: INVITE)
^M^[[K2850CLI> ^M^[[0Kset_destination: Parsing sip:201@EXTERNALIP:5060;transport=udp for address/port to send to
^M^[[K2850CLI> ^M^[[0Kset_destination: set destination to EXTERNALIP:5060
^M^[[K2850*CLI> ^M^[[0KReliably Transmitting (NAT) to EXTERNALIP:5060:
BYE sip:201@EXTERNALIP:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK67e237d4;rport^M
Max-Forwards: 70^M
From: sip:3581619@mydns.com;tag=as1eb0402a^M
To: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX 11.5.1^M
Proxy-Authorization: Digest username=“201”, realm=“asterisk”, algorithm=MD5, uri=“sip:mydns.com”, nonce=“2cced021”, response=“dada9a373eee1ad5fbced655886a9abc”^M
X-Asterisk-HangupCause: No user responding^M
X-Asterisk-HangupCauseCode: 18^M
Content-Length: 0^M
^M

^M^[[K2850*CLI> ^M^[[0KRetransmitting #1 (NAT) to EXTERNALIP:5060:
BYE sip:201@EXTERNALIP:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK67e237d4;rport^M
Max-Forwards: 70^M
From: sip:3581619@mydns.com;tag=as1eb0402a^M
To: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae^M
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX 11.5.1^M
Proxy-Authorization: Digest username=“201”, realm=“asterisk”, algorithm=MD5, uri=“sip:mydns.com”, nonce=“2cced021”, response=“dada9a373eee1ad5fbced655886a9abc”^M
X-Asterisk-HangupCause: No user responding^M
X-Asterisk-HangupCauseCode: 18^M
Content-Length: 0^M
^M

^M^[[K2850*CLI> ^M^[[0K
<— SIP read from UDP:EXTERNALIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK67e237d4;rport;received=72.200.89.91
From: sip:3581619@mydns.com;tag=as1eb0402a
To: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
CSeq: 102 BYE
Server: Cisco-CP7940G/8.0
Content-Length: 0

<------------->
— (8 headers 0 lines) —
^M^[[K2850CLI> ^M^[[0KSIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14’ Method: INVITE
^M^[[K2850CLI> ^M^[[0K
<— SIP read from UDP:EXTERNALIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK67e237d4;rport;received=72.200.89.91
From: sip:3581619@mydns.com;tag=as1eb0402a
To: “201” sip:201@mydns.com;tag=000b5faaaf9c02751850ff48-3f595aae
Call-ID: 000b5faa-af9c003f-2204ead7-480dde30@192.168.1.14
CSeq: 102 BYE
Server: Cisco-CP7940G/8.0
Content-Length: 0

<------------->
— (8 headers 0 lines) —
[/quote]

I replaced the IP coming from the phone outside of the network as “EXTERNALIP” and my DNS as “mydns.com”.
This is without STUN server. All required ports forwarded. Firewall disabled on server.