[asterisk-bugs] [Asterisk 0017896]: chan_multicast_rtp.so MulticastRTP no audio when using Page() App

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Aug 24 21:40:56 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17896 
====================================================================== 
Reported By:                svinson
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17896
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.0-beta3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-08-20 16:38 CDT
Last Modified:              2010-08-24 21:40 CDT
====================================================================== 
Summary:                    chan_multicast_rtp.so    MulticastRTP no audio when
using Page()  App
Description: 
When I use the Page() app with the MulticastRTP channel the phone answers
but i don't get any audio. when I use the Dial() command the audio works
fine. 
the Page() command works fine with the SIP channel. just not the
MulticastRTP channel. let me know what i can do to help debug this issue.
Thanks,
====================================================================== 

---------------------------------------------------------------------- 
 (0126295) svinson (reporter) - 2010-08-24 21:40
 https://issues.asterisk.org/view.php?id=17896#c126295 
---------------------------------------------------------------------- 
my sip.conf is defaults other than:

udpbindaddr=:: 

(and the phone config)

[4005]
context=default
type=friend
secret=4005
callerid="test" <4005>
host=dynamic                    
;nat=yes                         
;directmedia=no                  
disallow=all
;allow=gsm                      
allow=alaw


 extensions.conf:

[page]
 (not working)
exten => 4000,1,Answer
exten => 4000,n,Set(CALLERID(name)={Page:${CALLERID(name) })
exten => 4000,n,Set(TIMEOUT(digit)=5)
exten => 4000,n,SIPAddHeader(Call-Info: Answer-After=0)  ;  Grandstream,
Snoms
exten => 4000,n,SIPAddHeader(Alert-Info: info=alert-autoanswer)
;AASTRA/poly
exten => 4000,n,Page(MulticastRTP/basic/224.0.1.200:9999)
exten => 4000,n,Hangup

(works)
exten => 4200,1,Answer
exten => 4200,n,Set(CALLERID(name)={Page:${CALLERID(name) })
exten => 4200,n,Set(TIMEOUT(digit)=5)
exten => 4200,n,SIPAddHeader(Call-Info: Answer-After=0)  ;  Grandstream,
Snoms
exten => 4200,n,SIPAddHeader(Alert-Info: info=alert-autoanswer)
;AASTRA/poly
exten => 4200,n,Page(SIP/4005)
exten => 4200,n,Hangup

(works)
exten => 4300,1,Dial(MulticastRTP/basic/224.0.1.200:9999)
exten => 4300,n,Hangup


also take a look at the pcap files, in the file mrtp-dial-cmd.pcap you
will see data in the data field for the 224.0.1.200 multicast address. but
in the file mrtp-page-cmd.pcap using the page command the data filed is all
ff's


Thanks, for your help. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-24 21:40 svinson        Note Added: 0126295                          
======================================================================




More information about the asterisk-bugs mailing list