Asterisk call drop after 30 seconds

Question

I've installed Asterisk and made a call using Android Zoiper app. It successfully connects two users and hear sound, but call drops after 30 seconds.

asterisk logs

[Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4176 retrans_pkt: 
Retransmission timeout reached on transmission lPsW4atWG- for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4205 retrans_pkt: 
Hanging up call lPsW4atWG- - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (from-sip, 1000, 1) exited non-zero on 'SIP/2000-0000000a'

Sip.conf

[general]
 context=default                       ; Default context for incoming calls
 ;
 bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
 bindaddr=0.0.0.0           ; IP address to bind to (0.0.0.0 binds to all)
 ;
 disallow=all                    ; First disallow all codecs
 allow=gsm
 allow=ulaw                      ; Allow codecs in order of preference
 ;
 register => 12121111111:1234:11111111@sipauth.deltathree.com/1000


allow=g729
allow=alaw
srvlookup=no
canreinvite=no
directrtpsetup=no
trustpid=yes
sendrpid=yes
qualify=yes
callevents=yes
insecure=invite
pedantic=no
useragent=Glastender PBX
videosupport=no
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no

nat=yes
media_address = XXX.52.91.XXX ; server ip address

It looks like I need to change something on sip.conf, and tried different configs. It is not working yet.. Do you see any problems?

SIP logs

interface: eth0 (10.7.21.0/255.255.255.0)
filter: ( port 5060 ) and (ip or ip6)
#
U 2014/04/15 00:22:15.941072 XX.53.122.134:5060 -> 10.8.21.XX:5060
INVITE sip:1000@sipdomain.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;rport.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com.
CSeq: 20 INVITE.
Call-ID: wh8Ai1e~0c.
Max-Forwards: 70.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Content-Length: 280.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Contact: <sip:2000@XX.53.122.134>;+sip.instance="<urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516>".
.
v=0.
o=2000 274 59 IN IP4 192.168.0.38.
s=Talk.
c=IN IP4 192.168.0.38.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
m=video 9078 RTP/AVP 103 99.
a=rtpmap:103 VP8/90000.
a=rtpmap:99 MP4V-ES/90000.
a=fmtp:99 profile-level-id=3.

#
U 2014/04/15 00:22:15.945220 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Length: 0.
.

#
U 2014/04/15 00:22:15.951499 10.8.21.XX:5060 -> 223.XX.130.50:40764
INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK70816646;rport.
Max-Forwards: 70.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>.
Contact: <sip:2000@10.8.21.XX:5060>.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Mon, 14 Apr 2014 15:22:15 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1811076761 1811076761 IN IP4 192.168.0.38.
s=Asterisk PBX 11.8.1.
c=IN IP4 192.168.0.38.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 2014/04/15 00:22:16.045285 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: sip:1000@223.XX.130.50:40764.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
.

#
U 2014/04/15 00:22:16.445425 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

#
U 2014/04/15 00:22:16.447116 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Length: 0.
.

#
U 2014/04/15 00:22:16.838201 XX.53.122.134:5060 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:19.275720 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: <sip:1000@223.XX.130.50:40764>;+sip.instance="<urn:uuid:307db642-fa79-44d8-835f-15152558c31a>".
Content-Type: application/sdp.
Content-Length: 176.
.
v=0.
o=1000 3792 2294 IN IP4 223.XX.130.50.
s=Talk.
c=IN IP4 223.XX.130.50.
b=AS:380.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:19.276630 10.8.21.XX:5060 -> 223.XX.130.50:40764
ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK730c16dd;rport.
Max-Forwards: 70.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:19.276978 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:19.776861 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:20.778018 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:22.777522 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:25.139894 XX.53.122.134:32840 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:26.777002 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:30.179568 XX.53.122.134:55180 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:30.777462 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:34.777660 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:38.777721 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:42.777667 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:46.776449 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:46.927655 XX.53.122.134:5060 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:50.776948 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:1000@10.8.21.XX:5060>.
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:51.278124 10.8.21.XX:5060 -> XX.53.122.134:5060
INVITE sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK348a4dc2;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Contact: <sip:1000@10.8.21.XX:5060>.
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1836373944 1836373945 IN IP4 117.52.91.12.
s=Asterisk PBX 11.8.1.
c=IN IP4 117.52.91.12.
t=0 0.
m=audio 19152 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 2014/04/15 00:22:51.278285 10.8.21.XX:5060 -> 223.XX.130.50:40764
INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK59c0124b;rport.
Max-Forwards: 70.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1811076761 1811076762 IN IP4 117.52.91.12.
s=Asterisk PBX 11.8.1.
c=IN IP4 117.52.91.12.
t=0 0.
m=audio 15858 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 2014/04/15 00:22:51.344965 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
.

#
U 2014/04/15 00:22:51.355122 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: <sip:1000@223.XX.130.50:40764>;+sip.instance="<urn:uuid:307db642-fa79-44d8-835f-15152558c31a>".
Content-Type: application/sdp.
Content-Length: 176.
.
v=0.
o=1000 3792 2296 IN IP4 223.XX.130.50.
s=Talk.
c=IN IP4 223.XX.130.50.
b=AS:380.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:51.355539 10.8.21.XX:5060 -> 223.XX.130.50:40764
ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK144199ce;rport.
Max-Forwards: 70.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.355619 10.8.21.XX:5060 -> 223.XX.130.50:40764
BYE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK0ac3adc4;rport.
Max-Forwards: 70.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 104 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.408414 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
From: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
.

#
U 2014/04/15 00:22:51.408837 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
From: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: <sip:2000@XX.53.122.134>;+sip.instance="<urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516>".
Content-Type: application/sdp.
Content-Length: 170.
.
v=0.
o=2000 274 61 IN IP4 192.168.0.38.
s=Talk.
c=IN IP4 192.168.0.38.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:51.409343 10.8.21.XX:5060 -> XX.53.122.134:5060
ACK sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK04d7bdd5;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Contact: <sip:1000@10.8.21.XX:5060>.
Call-ID: wh8Ai1e~0c.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.409471 10.8.21.XX:5060 -> XX.53.122.134:5060
BYE sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK1b9de0d9;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.453121 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK0ac3adc4;rport.
From: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 104 BYE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

#
U 2014/04/15 00:22:51.495263 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK1b9de0d9;rport.
From: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 103 BYE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

exit
37 received, 0 dropped

Thank you.

Vivek Raj · Accepted Answer

This problem arises due to firewall and nating on server.you just have to follow following steps: 1) first of go through the firewall settings and check whether server's ip are white-listed there or not. 2)If you have already checked the above points then you are definitely facing NAT problem, To overcome this issue you have to add following parameters in sip.conf

[general]
externip=XXX.XX.91.XX
localnet=10.2.32.12/255.255.255.0
nat=yes

kaushik parmar · Answer

       UA1              Your Asterisk Server     UA2
      (IPv4)            (IPv4/IPv6)             (IPv6)
        |                    |                    |
        |   F1 INVITE        |                    |
        |------------------->|      F2 INVITE     |
        |                    |------------------->|
        |    100 Trying      |                    |
        |<-------------------|                    |
        |                    |    F3 200 OK       |
        |    F4 200 OK       |<-------------------|
        |<-------------------|                    |
        |                    |                    |
        |       F5 ACK       |                    |
        |------------------->|       F6 ACK       |
        |                    |------------------->|
        |                    |                    |
        |                    |        F7 BYE      |
        |       F8 BYE       |<-------------------|
        |<-------------------|                    |

Problem here is your UA1 is not getting ACK from second UA2. I had same problem and i came to know that every sip dialer has default 30 seconds of sip call timeout , so it hangup after 30 seconds as UA2 not received ACK signal. Post your full stack track by below command on cli so i can i help you to resolve this.

CLI> sip set debug on

I observed one reason behind this is NAT problem. your device is behind NAT and asterisk not able to send ACK signal to your registered device so it giving retransmission timeout for ACK signal.

Asterisk call drop after 30 seconds

Tags:

asterisk

Jake

2 Answers

Vivek Raj

kaushik parmar

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Asterisk call drop after 30 seconds

Tags:

asterisk

Jake

2 Answers

Vivek Raj

kaushik parmar

Related questions

Recent Activity

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