[Home]

Summary:ASTERISK-15592: Asterisk drop calls sending a CSeq: 103 BYE
Reporter:Jose Adan Tapia (voipsystems)Labels:
Date Opened:2010-02-08 15:03:48.000-0600Date Closed:2011-06-07 14:00:22
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I have a really big problem with the last versions of asterisk, apparently asterisk sends a BYE message between the first 5 and 20 minutes of outgoing an intbound calls, I put the log, I tried to add the ignoresdpversion=yes at my sip.conf but nothing happend, i patched my asterisk too with the patch in https://issues.asterisk.org/view.php?id=13958

I hope that something can help me, I use too the last version of asterisk 1.4 and dahdi complete but is the same issue.

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

<------------->
Scheduling destruction of SIP dialog '633f6a1071c558c777978e4462a122d4@192.168.1.3' in 7616 ms (Method: INVITE)
set_destination: Parsing <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b> for address/port to send to
set_destination: set destination to 192.168.1.6, port 56148
Reliably Transmitting (NAT) to 192.168.1.6:56148:
BYE sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK64915458;rport
Max-Forwards: 70
From: "24429643" <sip:24429643@192.168.1.3>;tag=as4ca504a5
To: <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b>;tag=aa167f7d
Call-ID: 633f6a1071c558c777978e4462a122d4@192.168.1.3
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
 == Spawn extension (default, s, 3) exited non-zero on 'DAHDI/2-1'
   -- Hungup 'DAHDI/2-1'
debian*CLI>
<--- SIP read from UDP:192.168.1.6:56148 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK64915458;rport=5060
Contact: <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b>
To: <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b>;tag=aa167f7d
From: "24429643"<sip:24429643@192.168.1.3>;tag=as4ca504a5
Call-ID: 633f6a1071c558c777978e4462a122d4@192.168.1.3
CSeq: 103 BYE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '633f6a1071c558c777978e4462a122d4@192.168.1.3' Method: INVITE
Reliably Transmitting (NAT) to 192.168.1.6:56148:
OPTIONS sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK7844621b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as5c5489d1
To: <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b>
Contact: <sip:asterisk@192.168.1.3>
Call-ID: 339b48685936185d443ff75e66a862d3@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.2
Date: Mon, 08 Feb 2010 20:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
debian*CLI>
<--- SIP read from UDP:192.168.1.6:56148 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK7844621b;rport=5060
Contact: <sip:192.168.1.6:56148>
To: <sip:1802@192.168.1.6:56148;rinstance=b569d3bd937c5e5b>;tag=fd1b2271
From: "asterisk"<sip:asterisk@192.168.1.3>;tag=as5c5489d1
Call-ID: 339b48685936185d443ff75e66a862d3@192.168.1.3
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '339b48685936185d443ff75e66a862d3@192.168.1.3' Method: OPTIONS
debian*CLI> core show channels verbose
Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode BridgedTo
0 active channels
0 active calls
3 calls processed
debian*CLI>
<--- SIP read from UDP:192.168.1.6:56148 --->
Comments:By: Leif Madsen (lmadsen) 2010-02-09 08:46:03.000-0600

You will need to provide the full SIP trace up to the BYE message as well -- just seeing the BYE message isn't terribly useful.

You should also provide the console output with debugging enabled, and SIP history enabled as well. This is per the bug guidelines when submitting a SIP issue.

Also, please attach as a text file to this issue instead of placing it inline. Thanks!
By: Leif Madsen (lmadsen) 2010-03-02 15:30:20.000-0600

In order to resolve this issue we'll need the information I've requested in my previous note. Without a response shortly we'll have to suspend this issue. Thanks!
By: Leif Madsen (lmadsen) 2010-03-17 10:45:50

Issue suspended due to lack of response from reporter.