[asterisk-bugs] [Asterisk 0019038]: Local Bridging not working

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19038 
====================================================================== 
Reported By:                cillinperera
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   19038
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.2.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-03-29 17:30 CDT
Last Modified:              2011-03-29 17:33 CDT
====================================================================== 
Summary:                    Local Bridging not working
Description: 
I seem to be having a very strange problem on 1.8.3.2. Bridging doesn't
seem to be working for a call which comes inbound on our DID numbers and is
then diverted to an extension, which is further instructed to dial an
outbound number. There is no ringtone, and no audio available.

However, dialing the extension directly works, as does inbound DID to a
SIP phone (or IVR).

This setup worked on our previous (1.4) installation.

extensions.conf

[from-reception]
include => ext-internal

[ext-internal]
exten => 107,1,Dial(SIP/sipcall/00353000000000,60)


sip.conf

[general]
port = 5060          
bindaddr = 0.0.0.0
disallow=all
allow=g729	
allow=ulaw
allow=alaw
srvlookup=yes
callerid = Unknown
alwaysauthreject=yes
dtmfmode=rfc2833

register => 41610000007:xxxxxx at myvoipprovider.com/107 

[sipcall]
type=peer
defaultuser=41610000000
secret=xxxxxxx
context=from-reception
host= myvoipprovider.com
fromuser=41615111100
qualify=yes
fromdomain=myvoipprovider.com
insecure=port,invite
caninvite=no
canreinvite=no
nat=no

that's literally it. I even stripped this right down, so these are the
only config files running. I get the following result:

 == Using SIP RTP CoS mark 5
    -- Executing [107 at from-reception:1] Dial("SIP/sipcall-00000023",
"SIP/sipcall/00353000000000,60") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sipcall/00353000000000
    -- SIP/sipcall-00000024 is making progress passing it to
SIP/sipcall-00000023
    -- SIP/sipcall-00000024 answered SIP/sipcall-00000023
    -- Locally bridging SIP/sipcall-00000023 and SIP/sipcall-00000024

but no audio. Have also tried using another provider on the outbound, no
luck.

this is pretty serious. We divert our incoming landlines to our mobiles
using this method, and nothing is working.
====================================================================== 

---------------------------------------------------------------------- 
 (0133169) cillinperera (reporter) - 2011-03-29 17:33
 https://issues.asterisk.org/view.php?id=19038#c133169 
---------------------------------------------------------------------- 
note that all of our other features are working with no issues, i.e. SIP to
PSTN, DID to SIP, Skype4Asterisk to PSTN, etc, etc, etc. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-29 17:33 cillinperera   Note Added: 0133169                          
======================================================================