[asterisk-bugs] [Asterisk 0013123]: segmentation fault with chan_h323 ast_rtp_new_source (rtp=0x0)

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13123 
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Reported By:                nik600
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13123
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             07-21-2008 09:56 CDT
Last Modified:              07-22-2008 04:49 CDT
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Summary:                    segmentation fault with chan_h323 ast_rtp_new_source
(rtp=0x0)
Description: 
i've installed asterisk 1.4.21.1
and configured with Open H.323 version v1.18.0, PWLib v1.10.0.

I've installed them but i'm experiencing this problem:

i've configured in h323.conf 2 peers:
one to an 3.3 CCM Cisco
one to an 4.2 CCM Cisco

each CCM has the preferred codec set up as G711 ulaw.

I can forward calls from a SIP account on asterisk (using Xten-xlite
as softphone) to both the peers and talk with their extensions without
any problem.

I can forward calls from both the peers to Asterisk (and for example
place the call in queue or background some sound files)

BUT

when i try to call from the CCM 3.3 to Asterisk, and then dial from
the dialplan a SIP account, when the SIP user accept the call (using
Xten-xlite as softphone) asterisk dies with a segmentation fault
error.

This happend only with CCM 3.3, with 4.2 there is no problem.

I've temporarly patched it editing chan_h323.c in the function
oh323_indicate at line 919 replacing 
// ********* OLD CODE
	case AST_CONTROL_SRCUPDATE:
		ast_rtp_new_source(pvt->rtp);
		res = 0;
		break;

// ***********
with
// ********* NEW CODE
	case AST_CONTROL_SRCUPDATE:
if(pvt->rtp!=NULL){
		ast_rtp_new_source(pvt->rtp);
}
		res = 0;
		break;

// ***********

with these "patch" asterisk doesn't crash.
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---------------------------------------------------------------------- 
 sergee - 07-22-08 04:49  
---------------------------------------------------------------------- 
that is proper patch, please upload it as a diff to ease life for bug
marshals, thanks. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-22-08 04:49  sergee         Note Added: 0090553                          
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