[asterisk-bugs] [Asterisk 0017795]: using chan_iax2 ringing received from destination sip channel doesn't always get propogated to originating sip

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=17795 
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Reported By:                jkroon
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17795
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.10 
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:             2010-08-04 10:46 CDT
Last Modified:              2010-08-04 10:46 CDT
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Summary:                    using chan_iax2 ringing received from destination
sip channel doesn't always get propogated to originating sip
Description: 
cell phone via pstn connects to broadcom infrastructure via some gateway,
so call gets asterisk server (1.6.2.10) via SIP.  This asterisk server then
passes call on to remote asterisk (1.6.2.9) server via chan_iax2 with
Dial(IAX2/peername/1234,,i).  The remote asterisk server then passes the
call to a handset using Dial(SIP/123).  At this point the calling agent can
correctly hear ringing and all channels are "down", so 123 answers and all
channels goes "up".  At this point when ext 123 (Snom 300 phone) hits the
transfer button the caller can hear MOH, and then when the new called party
gets notified (again hitting a Dial(SIP/321) in the dialplan) the calling
party however no longer hears ringing at all.

I have tested this with a SIP phone (snom 300) connected directly to the
first asterisk server as well, and this exhibits the same problem.

In the case where the calling party is coming in via broadcom on a second
transfer we do get a ringing sound, in the case where I connect the snom300
directly to the first asterisk server I don't.  (This part makes absolutely
no sense for me)

If I replace the IAX/2 channel with SIP it works as expected.  When the
call comes in directly on the second server from DAHDI/ it works.

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Issue History 
Date Modified    Username       Field                    Change               
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2010-08-04 10:46 jkroon         New Issue                                    
2010-08-04 10:46 jkroon         Asterisk Version          => 1.6.2.10        
2010-08-04 10:46 jkroon         Regression                => No              
2010-08-04 10:46 jkroon         SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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