Asterisk call hangup after 32 seconds

Hi all,
I am using asterisk as public server behind nat. I have two linphones (android and IOS) registered to it. when I try to make a call from android to IOS i get 2-way audio, however when I dial from IOS to android I get one way audio. In both cases call hangup after 32 seconds. I have made the call with debug mode on so you guys can have a look and suggest the possible solutions:
I have extension 2001 (android) and 2002 (IOS), in this example I made a call from extension 2001 to extension 2002:

my debug output is following:
Preformatted text
– Executing [2002@internal:1] NoOp(“SIP/2001-00000010”, “Dialing 2002”) in new stack
– Executing [2002@internal:2] Dial(“SIP/2001-00000010”, “SIP/2002,60”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 13172
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to x.x.x.xclients public ip):6335:
INVITE sip:2002@x.x.x.xclients public ip):6335;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK0dc8e2e5
Max-Forwards: 70
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>
Contact: sip:2001@192.168.68.3:5060
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1
Date: Wed, 26 Apr 2017 10:15:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1865732138 1865732138 IN IP4 192.168.66.32
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.32
t=0 0
m=audio 7076 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


-- Called SIP/2002

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK0dc8e2e5
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK0dc8e2e5
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 102 INVITE
User-Agent: Linphone_iPhone.6s.Plus_iOS10.3.1/3.16.2 (belle-sip/1.5.0)
Supported: replaces, outbound

<------------->
— (8 headers 0 lines) —
list_route: no route
– SIP/2002-00000011 is ringing

<— Transmitting (NAT) to x.x.x.xclients public ip):50512 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Length: 0

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

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK0dc8e2e5
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 102 INVITE
User-Agent: Linphone_iPhone.6s.Plus_iOS10.3.1/3.16.2 (belle-sip/1.5.0)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;+sip.instance="urn:uuid:596e01d4-f01f-45dd-adfb-f2e4e06aab2c"
Content-Type: application/sdp
Content-Length: 157

v=0
o=2002 2973 489 IN IP4 192.168.66.21
s=Talk
c=IN IP4 192.168.66.21
b=AS:380
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:102 telephone-event/8000
<------------->
— (12 headers 8 lines) —
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 102
Found audio description format telephone-event for ID 102
Capabilities: us - (gsm|ulaw|alaw|h263p|h264), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.66.21:7282
list_route: hop: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>
set_destination: Parsing <sip:2002@x.x.x.xclients public ip):6335;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):6335
Transmitting (no NAT) to x.x.x.xclients public ip):6335:
ACK sip:2002@x.x.x.xclients public ip):6335;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK2909b53d
Max-Forwards: 70
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Contact: sip:2001@192.168.68.3:5060
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0


-- SIP/2002-00000011 answered SIP/2001-00000010

Audio is at 14124
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to x.x.x.xclients public ip):50512 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv

<------------>
– Remotely bridging SIP/2001-00000010 and SIP/2002-00000011
Retransmitting #1 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Retransmitting #2 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Retransmitting #3 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Retransmitting #4 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Retransmitting #5 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:192.168.68.81:5060 —>
OPTIONS sip:192.168.68.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.68.81:5060;rport;branch=z9hG4bK1303837949
From: sip:2008@192.168.68.3;tag=712823039
To: sip:192.168.68.3
Call-ID: 1412397669
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.68.81:5060 (no NAT)
Looking for s in internal (domain 192.168.68.3)

<— Transmitting (no NAT) to 192.168.68.81:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.68.81:5060;branch=z9hG4bK1303837949;received=192.168.68.81;rport=5060
From: sip:2008@192.168.68.3;tag=712823039
To: sip:192.168.68.3;tag=as04b0eeeb
Call-ID: 1412397669
CSeq: 20 OPTIONS
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1412397669’ in 32000 ms (Method: OPTIONS)
Retransmitting #6 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Really destroying SIP dialog ‘1943050248’ Method: OPTIONS

<— SIP read from UDP:192.168.68.80:5060 —>
REGISTER sip:192.168.68.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.68.80:5060;rport;branch=z9hG4bK847314218
From: sip:2000@3cx;tag=499935301
To: sip:2000@3cx
Call-ID: 1438943716
CSeq: 655 REGISTER
Contact: sip:2000@192.168.68.80:5060;line=88de229c1433991
Authorization: Digest username=“2000”, realm=“asterisk”, nonce=“16d97f2a”, uri=“sip:192.168.68.3”, response=“cac262a76a8a810ff7557b24a1a88c48”, algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Expires: 600
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.68.80:5060 (no NAT)
Sending to 192.168.68.80:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.68.80:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.68.80:5060;branch=z9hG4bK847314218;received=192.168.68.80;rport=5060
From: sip:2000@3cx;tag=499935301
To: sip:2000@3cx;tag=as42e766b0
Call-ID: 1438943716
CSeq: 655 REGISTER
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="416553eb"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1438943716’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.68.80:5060 —>
REGISTER sip:192.168.68.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.68.80:5060;rport;branch=z9hG4bK1603569753
From: sip:2000@3cx;tag=499935301
To: sip:2000@3cx
Call-ID: 1438943716
CSeq: 656 REGISTER
Contact: sip:2000@192.168.68.80:5060;line=88de229c1433991
Authorization: Digest username=“2000”, realm=“asterisk”, nonce=“416553eb”, uri=“sip:192.168.68.3”, response=“35ab24732be352b3a2b8f13548bb62ae”, algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Expires: 600
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.68.80:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.68.80:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.68.80:5060;branch=z9hG4bK1603569753;received=192.168.68.80;rport=5060
From: sip:2000@3cx;tag=499935301
To: sip:2000@3cx;tag=as42e766b0
Call-ID: 1438943716
CSeq: 656 REGISTER
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: sip:2000@192.168.68.80:5060;line=88de229c1433991;expires=600
Date: Wed, 26 Apr 2017 10:15:48 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1438943716’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.68.80:5060 —>
OPTIONS sip:192.168.68.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.68.80:5060;rport;branch=z9hG4bK1260070316
From: sip:2000@192.168.68.3;tag=2099818488
To: sip:192.168.68.3
Call-ID: 131124555
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.68.80:5060 (no NAT)
Looking for s in internal (domain 192.168.68.3)

<— Transmitting (no NAT) to 192.168.68.80:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.68.80:5060;branch=z9hG4bK1260070316;received=192.168.68.80;rport=5060
From: sip:2000@192.168.68.3;tag=2099818488
To: sip:192.168.68.3;tag=as68dc1122
Call-ID: 131124555
CSeq: 20 OPTIONS
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘131124555’ in 32000 ms (Method: OPTIONS)
Retransmitting #7 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Really destroying SIP dialog ‘10614661’ Method: OPTIONS

<— SIP read from UDP:x.x.x.xclients public ip):50512 —>

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

<— SIP read from UDP:192.168.68.102:5060 —>
OPTIONS sip:192.168.68.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.68.102:5060;rport;branch=z9hG4bK640505046
From: sip:2005@192.168.68.3;tag=1081397053
To: sip:192.168.68.3
Call-ID: 1036142207
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.68.102:5060 (no NAT)
Looking for s in internal (domain 192.168.68.3)

<— Transmitting (no NAT) to 192.168.68.102:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.68.102:5060;branch=z9hG4bK640505046;received=192.168.68.102;rport=5060
From: sip:2005@192.168.68.3;tag=1081397053
To: sip:192.168.68.3;tag=as5314f67d
Call-ID: 1036142207
CSeq: 20 OPTIONS
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1036142207’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>

<------------->
Retransmitting #8 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Really destroying SIP dialog ‘668329711’ Method: OPTIONS
Retransmitting #9 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Retransmitting #10 (NAT) to x.x.x.xclients public ip):50512:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.66.32:41376;branch=z9hG4bK.pRetHIKqv;received=x.x.x.xclients public ip);rport=50512
From: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
To: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
Call-ID: mLJn-o8vfz
CSeq: 21 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2002@192.168.68.3:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1555288619 1555288619 IN IP4 192.168.66.21
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.66.21
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


[Apr 26 11:16:04] WARNING[3141]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission mLJn-o8vfz for seqno 21 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 26 11:16:04] WARNING[3141]: chan_sip.c:4067 retrans_pkt: Hanging up call mLJn-o8vfz - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
set_destination: Parsing <sip:2001@x.x.x.xclients public ip):50512;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):50512
Audio is at 14124
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to x.x.x.xclients public ip):50512:
INVITE sip:2001@x.x.x.xclients public ip):50512;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK18459054;rport
Max-Forwards: 70
From: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Contact: sip:2002@192.168.68.3:5060
Call-ID: mLJn-o8vfz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1555288619 1555288620 IN IP4 192.168.68.3
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.68.3
t=0 0
m=audio 14124 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


set_destination: Parsing <sip:2002@x.x.x.xclients public ip):6335;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):6335
Audio is at 13172
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to x.x.x.xclients public ip):6335:
INVITE sip:2002@x.x.x.xclients public ip):6335;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK00d462cd
Max-Forwards: 70
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Contact: sip:2001@192.168.68.3:5060
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1865732138 1865732139 IN IP4 192.168.68.3
s=Asterisk PBX 11.25.1
c=IN IP4 192.168.68.3
t=0 0
m=audio 13172 RTP/AVP 8 0 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv


Scheduling destruction of SIP dialog ‘7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060’ in 32000 ms (Method: INVITE)
== Spawn extension (internal, 2002, 2) exited non-zero on 'SIP/2001-00000010’
Scheduling destruction of SIP dialog ‘mLJn-o8vfz’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK00d462cd
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 103 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:x.x.x.xclients public ip):6335 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK00d462cd
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 103 INVITE
User-Agent: Linphone_iPhone.6s.Plus_iOS10.3.1/3.16.2 (belle-sip/1.5.0)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;+sip.instance="urn:uuid:596e01d4-f01f-45dd-adfb-f2e4e06aab2c"
Content-Type: application/sdp
Content-Length: 157

v=0
o=2002 2973 491 IN IP4 192.168.66.21
s=Talk
c=IN IP4 192.168.66.21
b=AS:380
t=0 0
m=audio 7282 RTP/AVP 8 0 102
a=rtpmap:102 telephone-event/8000
<------------->
— (12 headers 8 lines) —
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 102
Found audio description format telephone-event for ID 102
Capabilities: us - (gsm|ulaw|alaw|h263p|h264), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.66.21:7282
set_destination: Parsing <sip:2002@x.x.x.xclients public ip):6335;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):6335
Transmitting (no NAT) to x.x.x.xclients public ip):6335:
ACK sip:2002@x.x.x.xclients public ip):6335;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK68b14b8b
Max-Forwards: 70
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Contact: sip:2001@192.168.68.3:5060
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0


set_destination: Parsing <sip:2002@x.x.x.xclients public ip):6335;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):6335
Reliably Transmitting (no NAT) to x.x.x.xclients public ip):6335:
BYE sip:2002@x.x.x.xclients public ip):6335;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK67469d19
Max-Forwards: 70
From: sip:2001@192.168.68.3;tag=as40f38437
To: <sip:2002@x.x.x.xclients public ip):6335;transport=udp>;tag=8kX8~8F
Call-ID: 7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 11.25.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


Scheduling destruction of SIP dialog ‘7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:x.x.x.xclients public ip):50512 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK18459054;rport
From: <sip:2002@x.x.x.x(server public ip)>;tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Call-ID: mLJn-o8vfz
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:x.x.x.xclients public ip):50512 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK18459054;rport
From: <sip:2002@x.x.x.x(server public ip)>;tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Call-ID: mLJn-o8vfz
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/3.2.5 (belle-sip/1.6.1)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:2001@x.x.x.xclients public ip):50512;transport=udp>;+sip.instance="urn:uuid:5503aff0-e823-473a-a56d-6a4b444abf85"
Content-Type: application/sdp
Content-Length: 158

v=0
o=2001 1490 3498 IN IP4 192.168.66.32
s=Talk
c=IN IP4 192.168.66.32
b=AS:380
t=0 0
m=audio 7076 RTP/AVP 8 0 102
a=rtpmap:102 telephone-event/8000
<------------->
— (12 headers 8 lines) —
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 102
Found audio description format telephone-event for ID 102
Capabilities: us - (gsm|ulaw|alaw|h263p|h264), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.66.32:7076
set_destination: Parsing <sip:2001@x.x.x.xclients public ip):50512;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):50512
Transmitting (NAT) to x.x.x.xclients public ip):50512:
ACK sip:2001@x.x.x.xclients public ip):50512;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK5ec0b82c;rport
Max-Forwards: 70
From: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Contact: sip:2002@192.168.68.3:5060
Call-ID: mLJn-o8vfz
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0


set_destination: Parsing <sip:2001@x.x.x.xclients public ip):50512;transport=udp> for address/port to send to
set_destination: set destination to x.x.x.xclients public ip):50512
Reliably Transmitting (NAT) to x.x.x.xclients public ip):50512:
BYE sip:2001@x.x.x.xclients public ip):50512;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.68.3:5060;branch=z9hG4bK56a5f192;rport
Max-Forwards: 70
From: sip:2002@x.x.x.x(server public ip);tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Call-ID: mLJn-o8vfz
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.25.1
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, algorithm=MD5, uri=“sip:x.x.x.x(server public ip)”, nonce=“7208dee1”, response="b86c1bdb4c9db30c2a5a7f28db6343f6"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘7872c5a859a9a48b410b9fe9425df6ee@192.168.68.3:5060’ Method: INVITE

<— SIP read from UDP:x.x.x.xclients public ip):50512 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.68.3:5060;received=x.x.x.x(server public ip);branch=z9hG4bK56a5f192;rport
From: <sip:2002@x.x.x.x(server public ip)>;tag=as6ef0dacb
To: <sip:2001@x.x.x.x(server public ip)>;tag=1CftPdC6f
Call-ID: mLJn-o8vfz
CSeq: 103 BYE
User-Agent: LinphoneAndroid/3.2.5 (belle-sip/1.6.1)
Supported: replaces, outbound

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

Scheduling destruction of SIP dialog ‘1336964661’ in 32000 ms (Method: OPTIONS)
zain-PowerEdge-R220*CLI> sip set debug off
SIP Debugging Disabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Apr 26 11:20:57] NOTICE[3141][C-0000014c]: chan_sip.c:25902 handle_request_invite: Call from ‘’ (92.114.32.32:5071) to extension ‘900442086380542’ rejected because extension not found in context ‘internal’.
[Apr 26 11:21:29] WARNING[3141]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission 22e9aaa4ebf7172eb506675b24dfb0ad for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Apr 26 11:27:44] NOTICE[3141][C-0000014d]: chan_sip.c:25902 handle_request_invite: Call from ‘’ (92.114.32.32:5070) to extension ‘000442086380542’ rejected because extension not found in context ‘internal’.
[Apr 26 11:28:16] WARNING[3141]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission 140a6be3d171a739b0403b3eaf9e0e6d for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+RetransmissionsPreformatted text

This forum has a bug which makes it necessary to post logs as unformatted text. When you do that, you will probably find that the client’s Contact header is wrong.

hi david,
thanks for your prompt response, please can you tell me what is meant to post logs as unformatted text?

Use </> from the menu bar.