[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 UPDATED. 
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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:                   minor
Priority:                   normal
Status:                     acknowledged
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:              2010-08-05 16:18 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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2010-08-05 16:18 lmadsen        Description Updated                          
2010-08-05 16:18 lmadsen        Steps to Reproduce Updated                    
2010-08-05 16:18 lmadsen        Additional Information Updated                  
 
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