[asterisk-bugs] [Asterisk 0019087]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19087 
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Reported By:                satish_lx
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19087
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.3.2 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-04-08 11:20 CDT
Last Modified:              2011-04-11 08:37 CDT
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Summary:                    chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
Description: 
I have 1.8.3 without realtime config. here is my basic dialplan.

exten => s,1,Dial(${ARG2}&IAX2/${ARG1},20,t)
exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s,3,NoOp(${DIALSTATUS})
exten => s-NOANSWER,1,Voicemail(${ARG1},u)                      ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)                         ; If they
press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b)                          ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                             ; If they
press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1)                              ; Treat
anything else as no answer
exten => a,1,VoicemailMain(${ARG1})                             ; If they
press *, send the user into VoicemailMain
exten => s-CONGESTION,1,Congestion



When i am calling any extension and if that extension is not available i
got following WARNING and wired revers lookup IP address 0.0.29.200:5060. I
have only single IP configured for SIP bindaddress. 


shirley*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [7624 at from-sip:1] Macro("SIP/7527-000000bc",
"stdexten,7624,SIP/7624") in new stack
    -- Executing [s at macro-stdexten:1] Dial("SIP/7527-000000bc",
"SIP/7624&IAX2/7624,20,t") in new stack
  == Using SIP RTP CoS mark 5
[Apr  8 12:08:51] WARNING[15176]: acl.c:698 ast_ouraddrfor: Cannot
connect
[Apr  8 12:08:51] WARNING[15176]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
    -- Called 7624
    -- Called 7624
[Apr  8 12:08:51] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 12:08:52] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 12:08:54] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 12:08:55] NOTICE[13911]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
    -- IAX2/0.0.29.200:4569-3414 is circuit-busy
    -- Hungup 'IAX2/0.0.29.200:4569-3414'
[Apr  8 12:08:58] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7527-000000bc' in macro 'stdexten'
  == Spawn extension (from-sip, 7624, 1) exited non-zero on
'SIP/7527-000000bc'
[Apr  8 12:09:06] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 12:09:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 12:09:23] WARNING[13920]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
6ceaa0ab274f9ce30c1002283c74c20f at 172.30.1.47:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
shirley*CLI> exit


everything working but i want if my channel is not available it should say
(INVALID extension) This dial plan perfectly working with asterisk 1.2
version.

-Satish 
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---------------------------------------------------------------------- 
 (0133587) satish_lx (reporter) - 2011-04-11 08:37
 https://issues.asterisk.org/view.php?id=19087#c133587 
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I am sorry for that but do you have fix for this issue ? Why this is only
in 1.8.x version ? 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-11 08:37 satish_lx      Note Added: 0133587                          
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