Some calls are not getting disconnected properly. I also think this is why some of our calls get dropped. Here is the trixbox output:
[Aug 10 08:03:32] VERBOSE[7497] logger.c: – Called nexVortex/18001231234
[Aug 10 08:03:35] VERBOSE[7497] logger.c: – SIP/nexVortex-086572b0 is making progress passing it to SIP/206-b7381b88
[Aug 10 08:03:38] VERBOSE[7497] logger.c: – SIP/nexVortex-086572b0 answered SIP/206-b7381b88
[Aug 10 08:04:11] ERROR[3173] chan_sip.c: FONALITY: dialoglist is in an infinite loop: dialoglist=86becb8, dialog->next=86572b0, dialog=86becb8
then I manually close the connection
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/206-b7381b88”, “hangupcall,”) in new stack
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/206-b7381b88”, “1?skiprg”) in new stack
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Goto (macro-hangupcall,s,4)
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/206-b7381b88”, “1?skipblkvm”) in new stack
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Goto (macro-hangupcall,s,7)
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/206-b7381b88”, “1?theend”) in new stack
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Goto (macro-hangupcall,s,9)
[Aug 10 08:16:36] VERBOSE[7497] logger.c: – Executing [s@macro-hangupcall:9] Hangup(“SIP/206-b7381b88”, “”) in new stack
[Aug 10 08:16:36] VERBOSE[7497] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/206-b7381b88’ in macro ‘hangupcall’
[Aug 10 08:16:36] VERBOSE[7497] logger.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/206-b7381b88’
[Aug 10 08:16:36] VERBOSE[7497] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/206-b7381b88’ in macro ‘dialout-trunk’
[Aug 10 08:16:36] VERBOSE[7497] logger.c: == Spawn extension (from-internal, 819727590111, 5) exited non-zero on ‘SIP/206-b7381b88’
[Aug 10 08:16:36] VERBOSE[7499] logger.c: == End MixMonitor Recording SIP/206-b7381b88
[Aug 10 08:16:36] WARNING[3164] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Aug 10 08:17:08] WARNING[3173] chan_sip.c: Trying to destroy “72b722c05b04d75e232dccde601a33fb@68.193.163.10”, not found in dialog list?!?!
[Aug 10 08:17:08] WARNING[3173] chan_sip.c: Trying to destroy “1f21d13e-11c70e5d@192.168.1.113”, not found in dialog list?!?!
I contacted our provider, nexvortex and they did a SIP trace:
We pulled a SIP trace(copied below) and we were able to see a BYE message sent from our proxy to your Asterisk PBX. Your box gets the BYE but does not send a 200 OK instead sends us but 481 Call leg/transaction does not exist message and things that the call is gone. Please engage the Fonality support on this issue and feel free to pass them the SIP trace below:
U 2010/08/10 12:03:32.295148 68.193.163.10:60005 → my.ip.address:5060
INVITE sip:18001231234@my.ip.address SIP/2.0…Via: SIP/2.0/UDP 68.193.163.1
0:5060;branch=z9hG4bK76546180;rport…Max-Forwards: 70…From: "Steve
" sip:206@68.193.163.10;tag=as65ce9e2e…To: <sip:18001231234@my.ip.address
…Contact: sip:206@68.193.163.10…Call-ID: 72b722c05b04d75e232dccde
601a33fb@68.193.163.10…CSeq: 103 INVITE…User-Agent: c PBX 1.6.0.10
-FONCORE-r40…Proxy-Authorization: Digest username=“myusername”, realm=“nexv
ortex.com”, algorithm=MD5, uri=“sip:18001231234@my.ip.address”, nonce=“4c61
4140e3b083e227151869102e0253c3e2f489”, response=“1e30cd40919c3249a604af73c2
334e08”…Date: Tue, 10 Aug 2010 12:03:32 GMT…
U 2010/08/10 12:03:32.336429 my.ip.address:5060 → 68.193.163.10:60005
SIP/2.0 100 trying – your call is important to us…Via: SIP/2.0/UDP 68.193
.163.10:5060;branch=z9hG4bK76546180;rport=60005…From: “Steve”
sip:206@68.193.163.10;tag=as65ce9e2e…To: sip:18001231234@my.ip.address
…Call-ID: 72b722c05b04d75e232dccde601a33fb@68.193.163.10…CSeq: 103 INVITE
…Server: Sip EXpress router (0.9.6 (i386/linux))…Content-Length: 0…Warni
ng: 392 my.ip.address:5060 “Noisy feedback tells: pid=29263 req_src_ip=68.1
93.163.10 req_src_port=60005 in_uri=sip:18001231234@my.ip.address out_uri=s
ip:18001231234@67.16.110.141 via_cnt==1”…
U 2010/08/10 12:03:38.756578 my.ip.address:5060 → 68.193.163.10:60005
SIP/2.0 200 OK…Via: SIP/2.0/UDP 68.193.163.10:5060;branch=z9hG4bK76546180;
rport=60005…From: “Steve” sip:206@68.193.163.10;tag=as65ce9e
2e…To: sip:18001231234@my.ip.address;tag=gK079dd06d…Call-ID: 72b722c05b
04d75e232dccde601a33fb@68.193.163.10…CSeq: 103 INVITE…Record-Route: <sip:
18001231234@my.ip.address:5060;nat=yes;ftag=as65ce9e2e;lr=on>…Accept: appl
ication/sdp, application/isup, application/dtmf, application/dtmf-relay, m
ultipart/mixed…Contact: sip:18001231234@67.16.110.141:5060…Allow: INVIT
E,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,
OPTIONS…Require: timer…Supported: timer,replaces…
call connected, our Proxy sent you a 200 OK
U 2010/08/10 12:03:38.776062 68.193.163.10:60005 → my.ip.address:5060
ACK sip:18001231234@67.16.110.141:5060 SIP/2.0…Via: SIP/2.0/UDP 68.193.163
.10:5060;branch=z9hG4bK7b8ae69d;rport…Route: <sip:18001231234@my.ip.address
:5060;nat=yes;ftag=as65ce9e2e;lr=on>…Max-Forwards: 70…From: "Steve
" sip:206@68.193.163.10;tag=as65ce9e2e…To: <sip:18001231234@my.ip.address
;tag=gK079dd06d…Contact: sip:206@68.193.163.10…Call-ID: 72b722c
05b04d75e232dccde601a33fb@68.193.163.10…CSeq: 103 ACK…User-Agent: Asteris
k PBX 1.6.0.10-FONCORE-r40…Content-Length: 0…
the call was terminated from 18001231234 (PSTN connect) side
BYE sip:206@68.193.163.10:60005 SIP/2.0…Via: SIP/2.0/UDP 67.16.110.141:506
0;branch=z9hG4bK07B93f30e8dc3e4649c…From: sip:18001231234@my.ip.address;
tag=gK079dd06d…To: “Steve” sip:206@68.193.163.10;tag=as65ce9
e2e…Call-ID: 72b722c05b04d75e232dccde601a33fb@68.193.163.10…CSeq: 16465 B
YE…Max-Forwards: 70…Route: <sip:18001231234@my.ip.address:5060;nat=yes;ft
ag=as65ce9e2e;lr=on>…Content-Length: 0…
U 2010/08/10 12:05:51.203200 my.ip.address:5060 → 68.193.163.10:60005
BYE sip:206@68.193.163.10:60005 SIP/2.0…Record-Route: <sip:206@my.ip.address
:5060;nat=yes;ftag=gK079dd06d;lr=on>…Via: SIP/2.0/UDP my.ip.address:5060
;branch=z9hG4bK6fd8.076d37a7.0…Via: SIP/2.0/UDP 67.16.110.141:5060;branch=
z9hG4bK07B93f30e8dc3e4649c…From: sip:18001231234@my.ip.address;tag=gK079
dd06d…To: “Steve” sip:206@68.193.163.10;tag=as65ce9e2e…Call
-ID: 72b722c05b04d75e232dccde601a33fb@68.193.163.10…CSeq: 16465 BYE…Max-F
orwards: 16…Content-Length: 0…
BYE message was sent from our proxy server to your Asterisk PBX
U 2010/08/10 12:05:51.224074 68.193.163.10:60005 → my.ip.address:5060
SIP/2.0 481 Call leg/transaction does not exist…Via: SIP/2.0/UDP my.ipaddress
:5060;branch=z9hG4bK6fd8.076d37a7.0;received=my.ip.address…Via: SIP/2.
0/UDP 67.16.110.141:5060;branch=z9hG4bK07B93f30e8dc3e4649c…From: <sip:1972
7590111@my.ip.address>;tag=gK079dd06d…To: “Steve” <sip:206@68.
193.163.10>;tag=as65ce9e2e…Call-ID: 72b722c05b04d75e232dccde601a33fb@68.19
3.163.10…CSeq: 16465 BYE…User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40…A
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY…Supporte
d: replaces, timer…Content-Length: 0…
your system responded with 481 Call leg/transaction does not exist
U 2010/08/10 12:16:36.319892 68.193.163.10:60005 → my.ip.address:5060
BYE sip:18001231234@67.16.110.141:5060 SIP/2.0…Via: SIP/2.0/UDP 68.193.163
.10:5060;branch=z9hG4bK02aab9fc;rport…Route: <sip:18001231234@my.ip.address
:5060;nat=yes;ftag=as65ce9e2e;lr=on>…Max-Forwards: 70…From: "Steve
" sip:206@68.193.163.10;tag=as65ce9e2e…To: <sip:18001231234@my.ip.address
;tag=gK079dd06d…Call-ID: 72b722c05b04d75e232dccde601a33fb@68.193.1
63.10…CSeq: 104 BYE…User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40…Proxy-
Authorization: Digest username=“myusername”, realm=“nexvortex.com”, algorith
m=MD5, uri=“sip:18001231234@67.16.110.141:5060”, nonce=“4c614140e3b083e2271
51869102e0253c3e2f489”, response=“bb7a45ca01965fc86dc8470dd0e39c9d”…X-Aste
risk-HangupCause: Normal Clearing…X-Asterisk-HangupCauseCode: 16…Content-
Length: 0…
this is when after manually disconnect the call
I posted on the trixbox forums without any response. Please let me know if anyone can help.
Thanks