[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 10:35:07 CST 2007


A NOTE has been added to this issue. 
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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 10:35 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 10:35  
---------------------------------------------------------------------- 
This patch is for asterisk version 1.4.5.  But for version 1.4.14, I put
this patch on line 15050 in chan_sip.c:

   if ((p->icseq && (p->icseq > seqno)) && !(!ast_test_flag(&p->flags[0],
SIP_OUTGOING) && (req->method == SIP_ACK) && p->pendinginvite &&
(p->pendinginvite == seqno))) {
		if (option_debug)
		    ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting
>= %d)\n", seqno, p->icseq);
		if (req->method != SIP_ACK)
			transmit_response(p, "503 Server error", req);	/* We must respond
according to RFC 3261 sec 12.2 */
		return -1;
	} else if (p->icseq &&
		   p->icseq == seqno &&
		   req->method != SIP_ACK &&
		   (p->method != SIP_CANCEL || ast_test_flag(&p->flags[0],
SIP_ALREADYGONE))) { 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-29-07 10:35  andrewgray     Note Added: 0074553                          
======================================================================




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