[asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Nov 29 10:27:59 CST 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11389 
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Reported By:                andrewgray
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11389
Category:                   Applications/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:            1.2.24  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             11-27-2007 12:47 CST
Last Modified:              11-29-2007 10:27 CST
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Summary:                    While you are dialing from ATA/listening to the ring
, if you press touchtone, then hangup,  the other line rings forever.
Description: 
Verified on Asterisk 1.2.24, 1.2.14, 1.2.13, 1.2.9.

Using an ATA, if you dial another extension, and while you are listening
to it ring, if you press a touchtone key sending out a DTMF tone, then hang
up your phone, the other phone rings forever.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0005215 [patch] Provisional responses to INVITE...
====================================================================== 

---------------------------------------------------------------------- 
 andrewgray - 11-29-07 10:27  
---------------------------------------------------------------------- 
OK.  Here is what issue 5215 said was the patch:

Asterisk also shouldn't ignore ACKs for the initial INVITE in a dialog
that it has received requests for in the early dialog. The following patch
works for me, but might let some requests through that should be ignored.


@@ -10329,7 +10330,8 @@
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP
%s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);

- if (p->icseq && (p->icseq > seqno)) {
+ if ((p->icseq && (p->icseq > seqno))
+ && !(!ast_test_flag(p, SIP_OUTGOING) && (req->method == SIP_ACK) &&
p->pendinginvite && (p->pendinginvite == seqno))) {
                ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d
(expecting >= %d)\n", seqno, p->icseq);
                return -1;
        } else if (p->icseq && (p->icseq == seqno) && req->method !=
SIP_ACK &&(p->method != SIP_CANCEL|| ast_test_flag(p, SIP_ALREADYGONE))) { 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-29-07 10:27  andrewgray     Note Added: 0074552                          
======================================================================




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