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Summary:ASTERISK-07079: SIP Channel Lock issue
Reporter:Christopher Chenoweth (cdyne)Labels:
Date Opened:2006-06-01 17:54:39Date Closed:2006-06-06 17:46:18
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) CantMatchBye.txt
Description:I get this error sometimes:

Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!

****** ADDITIONAL INFORMATION ******

Here is all the SIP Debug I have:


-----------------------------  GOOD ONE ---------------------------------------------------
We're at 70.85.7.246 port 10234
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 4.79.212.236:5060:
INVITE sip:+17575125254@4.79.212.236 SIP/2.0
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK67f880c1;rport
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
To: <sip:+17575125254@4.79.212.236>
Contact: <sip:0004110000@70.85.7.246>
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Thu, 01 Jun 2006 22:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 30274 30274 IN IP4 70.85.7.246
s=session
c=IN IP4 70.85.7.246
t=0 0
m=audio 10234 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK67f880c1;rport=5060
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
To: <sip:+17575125254@4.79.212.236>
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 102 INVITE
Server: Bandwidth.com SDE (v1.0)
Content-Length: 0
Warning: 392 4.79.212.234:5060 "Noisy feedback tells:  pid=32355 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17575125254@4.79.212.236 out_uri=sip:+17575125254@4.68.250.148:5060 via_cnt==2"


--- (9 headers 0 lines)---
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK67f880c1;rport=5060
Record-Route: <sip:4.79.212.236;lr;ftag=as026b968a>,<sip:4.79.212.234;lr;ftag=as026b968a>,<sip:4.79.212.236;lr;ftag=as026b968a>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 102 INVITE
Contact: <sip:+17575125254@4.68.250.148:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 182

v=0
o=- 1149200648 1149200649 IN IP4 209.247.5.231
s=-
c=IN IP4 209.247.5.231
t=0 0
m=audio 60112 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--- (10 headers 9 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.247.5.231:60112
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK67f880c1;rport=5060
Record-Route: <sip:4.79.212.236;lr;ftag=as026b968a>,<sip:4.79.212.234;lr;ftag=as026b968a>,<sip:4.79.212.236;lr;ftag=as026b968a>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 102 INVITE
Contact: <sip:+17575125254@4.68.250.148:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 182

v=0
o=- 1149200648 1149200649 IN IP4 209.247.5.231
s=-
c=IN IP4 209.247.5.231
t=0 0
m=audio 60112 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--- (10 headers 9 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.247.5.231:60112
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:4.79.212.236;lr;ftag=as026b968a>
list_route: hop: <sip:4.79.212.234;lr;ftag=as026b968a>
list_route: hop: <sip:4.79.212.236;lr;ftag=as026b968a>
set_destination: Parsing <sip:4.79.212.236;lr;ftag=as026b968a> for address/port to send to
set_destination: set destination to 4.79.212.236, port 5060
Transmitting (no NAT) to 4.79.212.236:5060:
ACK sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK6dbc6659;rport
Route: <sip:4.79.212.236;lr;ftag=as026b968a>,<sip:4.79.212.234;lr;ftag=as026b968a>,<sip:4.79.212.236;lr;ftag=as026b968a>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Contact: <sip:0004110000@70.85.7.246>
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0


---
Jun  1 18:24:13 NOTICE[30564]: app_amd.c:140 isAnsweringMachine: AMD using the default parameters.
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bKcde1.d8eb0e12.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497889828
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Sending to 4.79.212.236 : 5060 (no NAT)
Transmitting (no NAT) to 4.79.212.236:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.79.212.236;branch=0;received=4.79.212.236
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bKcde1.d8eb0e12.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497889828
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as026b968a
Call-ID: 3e44080241aa60913e9d3bb3208638c3@70.85.7.246
CSeq: 1 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0004110000@70.85.7.246>
Content-Length: 0


---
Really destroying SIP dialog '3e44080241aa60913e9d3bb3208638c3@70.85.7.246' Method: BYE

------------------------------------------------------------------------------------------------------
Same Call.. went bad...
------------------------------------------------------------------------------------------------------

We're at 70.85.7.246 port 15300
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 4.79.212.236:5060:
INVITE sip:+17575125254@4.79.212.236 SIP/2.0
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK0980dd83;rport
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>
Contact: <sip:0004110000@70.85.7.246>
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Thu, 01 Jun 2006 22:24:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 30274 30274 IN IP4 70.85.7.246
s=session
c=IN IP4 70.85.7.246
t=0 0
m=audio 15300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK0980dd83;rport=5060
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 102 INVITE
Server: Bandwidth.com SDE (v1.0)
Content-Length: 0
Warning: 392 4.79.212.234:5060 "Noisy feedback tells:  pid=32364 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17575125254@4.79.212.236 out_uri=sip:+17575125254@4.68.250.148:5060 via_cnt==2"


--- (9 headers 0 lines)---
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK0980dd83;rport=5060
Record-Route: <sip:4.79.212.236;lr;ftag=as27eb4ac5>,<sip:4.79.212.234;lr;ftag=as27eb4ac5>,<sip:4.79.212.236;lr;ftag=as27eb4ac5>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 102 INVITE
Contact: <sip:+17575125254@4.68.250.148:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 182

v=0
o=- 1149200669 1149200670 IN IP4 209.247.23.73
s=-
c=IN IP4 209.247.23.73
t=0 0
m=audio 61134 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--- (10 headers 9 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.247.23.73:61134
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK0980dd83;rport=5060
Record-Route: <sip:4.79.212.236;lr;ftag=as27eb4ac5>,<sip:4.79.212.234;lr;ftag=as27eb4ac5>,<sip:4.79.212.236;lr;ftag=as27eb4ac5>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 102 INVITE
Contact: <sip:+17575125254@4.68.250.148:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 182

v=0
o=- 1149200669 1149200670 IN IP4 209.247.23.73
s=-
c=IN IP4 209.247.23.73
t=0 0
m=audio 61134 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--- (10 headers 9 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.247.23.73:61134
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:4.79.212.236;lr;ftag=as27eb4ac5>
list_route: hop: <sip:4.79.212.234;lr;ftag=as27eb4ac5>
list_route: hop: <sip:4.79.212.236;lr;ftag=as27eb4ac5>
set_destination: Parsing <sip:4.79.212.236;lr;ftag=as27eb4ac5> for address/port to send to
set_destination: set destination to 4.79.212.236, port 5060
Transmitting (no NAT) to 4.79.212.236:5060:
ACK sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK3af1be87;rport
Route: <sip:4.79.212.236;lr;ftag=as27eb4ac5>,<sip:4.79.212.234;lr;ftag=as27eb4ac5>,<sip:4.79.212.236;lr;ftag=as27eb4ac5>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Contact: <sip:0004110000@70.85.7.246>
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0


---
Jun  1 18:24:35 NOTICE[30568]: app_amd.c:140 isAnsweringMachine: AMD using the default parameters.
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890sip821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:24:50 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:24:50 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:24:50 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:24:51 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:24:51 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:24:51 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:24:53 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:24:53 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:24:53 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:24:57 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:24:57 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:24:57 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:25:01 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:01 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:01 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:25:05 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:05 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:05 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:25:09 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:09 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:09 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:25:13 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:13 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:13 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
BYE sip:0004110000@70.85.7.246 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.234;lr;ftag=VPST506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPST506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK44af.b3f45721.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1138497890821
From: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
To: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 1 BYE
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0


--- (15 headers 0 lines)---
Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13079 sipsock_read: We could NOT get the channel lock for SIP/level3-6920!
Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13080 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Jun  1 18:25:17 ERROR[30298]: chan_sip.c:13081 sipsock_read: BAD! BAD! BAD!
set_destination: Parsing <sip:4.79.212.236;lr;ftag=as27eb4ac5> for address/port to send to
set_destination: set destination to 4.79.212.236, port 5060
Reliably Transmitting (no NAT) to 4.79.212.236:5060:
BYE sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK01289cb1;rport
Route: <sip:4.79.212.236;lr;ftag=as27eb4ac5>,<sip:4.79.212.234;lr;ftag=as27eb4ac5>,<sip:4.79.212.236;lr;ftag=as27eb4ac5>
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Contact: <sip:0004110000@70.85.7.246>
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 103 BYE
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0


---
ast2*CLI>
<-- SIP read from 4.79.212.236:5060:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK01289cb1;rport=5060
From: "0004110000" <sip:0004110000@70.85.7.246>;tag=as27eb4ac5
To: <sip:+17575125254@4.79.212.236>;tag=VPST506071629460
Call-ID: 6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246
CSeq: 103 BYE
Content-Length: 0


--- (7 headers 0 lines)---
Jun  1 18:25:56 WARNING[30298]: chan_sip.c:10758 handle_response: Remote host can't match request BYE to call '6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246'. Giving up.
Really destroying SIP dialog '6973c67e0db1d6ff2bf19f9b6d550d9e@70.85.7.246' Method: INVITE
ast2*CLI>
Comments:By: Serge Vecher (serge-v) 2006-06-01 18:18:22

1) I see app_amd in use from the the log. Afaik, it is not 1.2.8, but only in trunk. So which version of Asterisk are you really using?
2) In what context does this happen? Can you narrow it down to a specific chain of events?
3) Please always attach logs as an _attachment_, not inline. Thanks
By: Christopher Chenoweth (cdyne) 2006-06-01 18:40:54

here is what I did.
cd /usr/src
rm -d -r -f asterisk (removed all of asterisk)
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk

I did that two hours ago.  I do an originate from the manager.  It works most of the time, but, others (I am playing wav files) don't work properly.

Sry about not uploading it via file.  :)
By: Christopher Chenoweth (cdyne) 2006-06-01 20:39:37

Hello.  I just noticed I put 1.2.8.  Sorry about that.  Yes.  It is SVN.

Long day...
By: Russell Bryant (russell) 2006-06-06 00:27:48

I believe I have fixed this problem in revision 32478.  Can you please retry and see if you're still having the problem?
By: Hadley Rich (hads) 2006-06-06 16:52:19

I reported this same problem on #asterisk-dev and russellb helped me out. Making calls normally was fine, this only happened with calls originated from the manager interface.

Was using r32456 when I was having problems and r32478 fixed the problem consistently.

Cheers.
By: Christopher Chenoweth (cdyne) 2006-06-06 17:42:54

Looks good to me.  I slammed some calls on the system.  And no issues.  Have a transfer problem now.  I will open another ticket for that.

Thanks,
Chris
By: Christopher Chenoweth (cdyne) 2006-06-06 17:44:42

BTW:  Thanks!!!
By: Russell Bryant (russell) 2006-06-06 17:46:18

You are very welcome.  I'm glad I was able to help.  :)