Asterisk call drop after 30 seconds -
i've installed asterisk , made call using android zoiper app. connects 2 users , hear sound, 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- seqno 20 (critical response) -- see https://wiki.asterisk.org/wiki/display/ast/sip+retransmissions packet timed out after 31999ms no response [apr 14 18:40:34] warning[27959]: chan_sip.c:4205 retrans_pkt: hanging call lpsw4atwg- - no reply 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 incoming calls ; bindport=5060 ; bindport local udp port asterisk listen on bindaddr=0.0.0.0 ; ip address bind (0.0.0.0 binds all) ; disallow=all ; first disallow 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 need change on sip.conf, , tried different configs. not working yet.. see problems?
sip logs
interface: eth0 (10.7.21.0/255.255.255.0) filter: ( port 5060 ) , (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.
this problem arises due firewall , nating on server.you have follow following steps: 1) first of go through firewall settings , check whether server's ip white-listed there or not. 2)if have checked above points facing nat problem, overcome issue have add following parameters in sip.conf
[general] externip=xxx.xx.91.xx localnet=10.2.32.12/255.255.255.0 nat=yes
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