Remote extensions call drops after 20 seconds

Asterisk Asterisk General
1

Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds.

I got my NAT firewall properly configured. Here I attached my debug in CLI:

[code]
e[0KReliably Transmitting (NAT) to 190.113.109.189:1024:
OPTIONS sip:500@190.113.109.189:5060 SIP/2.0
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK580703b1;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@201.237.180.158;tag=as7de0aab0
To: sip:500@190.113.109.189:5060
Contact: sip:Unknown@201.237.180.158:5060
Call-ID: 4e00bf825f48c9fd06a1522f07b0f84f@201.237.180.158:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 18 Dec 2013 17:55:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:190.113.109.189:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK580703b1;rport=5060
From: “Unknown” sip:Unknown@201.237.180.158;tag=as7de0aab0
To: sip:500@190.113.109.189:5060;tag=1444883487
Call-ID: 4e00bf825f48c9fd06a1522f07b0f84f@201.237.180.158:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.5.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->

<— SIP read from UDP:190.113.109.189:1024 —>
INVITE sip:83226044@201.237.180.158 SIP/2.0
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK291986337;rport
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 180 INVITE
Contact: sip:500@190.113.109.189:5060
Max-Forwards: 70
User-Agent: Grandstream GXP1160 1.0.5.15
Privacy: none
P-Preferred-Identity: sip:500@201.237.180.158
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 383

v=0
o=500 8000 8000 IN IP4 190.113.109.189
s=SIP Call
c=IN IP4 190.113.109.189
t=0 0
m=audio 5004 RTP/AVP 18 8 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->

<— Reliably Transmitting (NAT) to 190.113.109.189:1024 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK291986337;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as078d7da9
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 180 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="46a6d67d"
Content-Length: 0

<------------>
e[0KRetransmitting #1 (NAT) to 190.113.109.189:1024:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK291986337;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as078d7da9
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 180 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="46a6d67d"
Content-Length: 0

<— SIP read from UDP:190.113.109.189:1024 —>
ACK sip:83226044@201.237.180.158 SIP/2.0
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK291986337;rport
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as078d7da9
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 180 ACK
Content-Length: 0
<------------->

<— SIP read from UDP:190.113.109.189:1024 —>
INVITE sip:83226044@201.237.180.158 SIP/2.0
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;rport
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Contact: sip:500@190.113.109.189:5060
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“46a6d67d”, uri="sip:83226044@201.237.180.158", response=“77056a42de69524617d73b3b9e55dda9”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1160 1.0.5.15
Privacy: none
P-Preferred-Identity: sip:500@201.237.180.158
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 383

v=0
o=500 8000 8000 IN IP4 190.113.109.189
s=SIP Call
c=IN IP4 190.113.109.189
t=0 0
m=audio 5004 RTP/AVP 18 8 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->

<— Transmitting (NAT) to 190.113.109.189:1024 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
Content-Length: 0

<------------>

<— Transmitting (NAT) to 190.113.109.189:1024 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287715 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

e[0K
<— SIP read from UDP:190.113.109.189:1024 —>
ACK sip:83226044@201.237.180.158 SIP/2.0
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK291986337;rport
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as078d7da9
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 180 ACK
Content-Length: 0

<------------->

<— Reliably Transmitting (NAT) to 190.113.109.189:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

e[0K – Locally bridging SIP/500-000000af and SIP/gateway-000000b0

e[0KRetransmitting #1 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KRetransmitting #2 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KRetransmitting #3 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KRetransmitting #4 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KRetransmitting #5 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KRetransmitting #6 (NAT) to 190.113.109.189:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK895970443;received=190.113.109.189;rport=1024
From: sip:500@201.237.180.158;tag=1785161547
To: sip:83226044@201.237.180.158;tag=as66b6dcff
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 181 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060

e[0KContent-Type: application/sdp
Content-Length: 314

v=0
o=root 2030287715 2030287716 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 18334 RTP/AVP 9 18 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

e[0KReliably Transmitting (NAT) to 190.113.109.189:1024:
BYE sip:500@190.113.109.189:5060 SIP/2.0
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK69adf13f;rport
Max-Forwards: 70
From: sip:83226044@201.237.180.158;tag=as66b6dcff
To: sip:500@201.237.180.158;tag=1785161547
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
Proxy-Authorization: Digest username=“500”, realm=“asterisk”, algorithm=MD5, uri=“sip:201.237.180.158”, nonce="", response="62d27937fdf6d838f8331e598518d361"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

e[0KRetransmitting #1 (NAT) to 190.113.109.189:1024:
BYE sip:500@190.113.109.189:5060 SIP/2.0
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK69adf13f;rport
Max-Forwards: 70
From: sip:83226044@201.237.180.158;tag=as66b6dcff
To: sip:500@201.237.180.158;tag=1785161547
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
Proxy-Authorization: Digest username=“500”, realm=“asterisk”, algorithm=MD5, uri=“sip:201.237.180.158”, nonce="", response="62d27937fdf6d838f8331e598518d361"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

<— SIP read from UDP:190.113.109.189:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK69adf13f;rport=5060
From: sip:83226044@201.237.180.158;tag=as66b6dcff
To: sip:500@201.237.180.158;tag=1785161547
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 102 BYE
Contact: sip:500@190.113.109.189:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.5.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->

e[0KSIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘599413041-5060-19@BJC.BGI.A.BAG’ Method: INVITE

<— SIP read from UDP:190.113.109.189:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK69adf13f;rport=5060
From: sip:83226044@201.237.180.158;tag=as66b6dcff
To: sip:500@201.237.180.158;tag=1785161547
Call-ID: 599413041-5060-19@BJC.BGI.A.BAG
CSeq: 102 BYE
Contact: sip:500@190.113.109.189:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.5.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->[/code]

Thank you!

2

You are not getting a reply to SIP OK from the remote server. When Asterisk sends a SIP OK (answer the call), the remote server should send SIP ACK back as conformation. There might be two reasons for that:

  • the VoIP provider is not sending the SIP ACK (highly unlikely)
  • the SIP ACK is blocked by the NAT router (or getting stuck somewhere in the path)
3

Hi. I still have this issue, but when I use Zoiper on remote, works fine. So, when remote person switch to Grandstream SIP Phone, call drops immediately.

Server is behind NAT. I have configured NAT in my iptables.

Any ideas? Thank you very much. This is my updated logs:

[code]Audio is at 20148
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 190.113.109.189:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912795 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– SIP/gateway-00000071 answered SIP/500-00000070
Audio is at 20148
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 190.113.109.189:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– Locally bridging SIP/500-00000070 and SIP/gateway-00000071
Retransmitting #1 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #2 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #3 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Registered SIP '500' at 190.113.109.189:35225
   > Saved useragent "Zoiper r21367" for peer 500

Retransmitting #4 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #5 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #6 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #6 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #8 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #9 (NAT) to 190.113.109.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
From: sip:500@201.237.180.158;tag=1826651759
To: sip:83226044@201.237.180.158;tag=as1776630f
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 21 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:83226044@201.237.180.158:5060
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 422912795 422912796 IN IP4 201.237.180.158
s=Asterisk PBX 1.8.20.0
c=IN IP4 201.237.180.158
t=0 0
m=audio 20148 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Scheduling destruction of SIP dialog ‘705414843-5060-3@BJC.BGI.A.BAB’ in 27968 ms (Method: INVITE)
set_destination: Parsing sip:500@190.113.109.189:5060 for address/port to send to
set_destination: set destination to 190.113.109.189:5060
Reliably Transmitting (NAT) to 190.113.109.189:5060:
BYE sip:500@190.113.109.189:5060 SIP/2.0
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK1e937f44;rport
Max-Forwards: 70
From: sip:83226044@201.237.180.158;tag=as1776630f
To: sip:500@201.237.180.158;tag=1826651759
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
Proxy-Authorization: Digest username=“500”, realm=“asterisk”, algorithm=MD5, uri=“sip:201.237.180.158”, nonce="", response="62d27937fdf6d838f8331e598518d361"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

Retransmitting #1 (NAT) to 190.113.109.189:5060:
BYE sip:500@190.113.109.189:5060 SIP/2.0
Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK1e937f44;rport
Max-Forwards: 70
From: sip:83226044@201.237.180.158;tag=as1776630f
To: sip:500@201.237.180.158;tag=1826651759
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
Proxy-Authorization: Digest username=“500”, realm=“asterisk”, algorithm=MD5, uri=“sip:201.237.180.158”, nonce="", response="62d27937fdf6d838f8331e598518d361"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

<— SIP read from UDP:190.113.109.189:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1e937f44;rport=5060
From: sip:83226044@201.237.180.158;tag=as1776630f
To: sip:500@201.237.180.158;tag=1826651759
Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
CSeq: 102 BYE
Contact: sip:500@190.113.109.189:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.5.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0[/code]

4

The SIP protocol is based on requests and replies. Both sides send requests and wait for replies. Some of these requests are important. In a TCP/IP network many things can happen with IP packets. Firewalls, NAT devices, Session Border Controllers and SIP Proxys are in the signalling path and they will affect the call.

For some reason signalling doesn’t work as expected between your Asterisk server and the other device. There could be many reasons why this happens.

A NAT device in the signalling path. A misconfigured NAT device is in the signalling path and stops SIP messages.
A firewall that blocks messages or reroutes them wrongly in an attempt to assist in a too clever way.
A SIP middlebox (SBC) that rewrites contact: headers so that we can't reach the other side with our reply or the ACK.
A badly configured SIP proxy that forgets to add record-route headers to make sure that signalling works.
Packet loss. IP and UDP are unreliable transports. If you loose too many packets the retransmits doesn't help and communication is impossible. If this happens with signaling, media would be unusable anyway.