[asterisk-bugs] [Asterisk 0010009]: "ast_bridge_call: Bridge failed" after ChannelRedirect

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Aug 30 19:32:20 CDT 2007


The following issue has been REOPENED. 
====================================================================== 
http://bugs.digium.com/view.php?id=10009 
====================================================================== 
Reported By:                dimas
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   10009
Category:                   Core/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 70163 
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             06-19-2007 18:58 CDT
Last Modified:              08-30-2007 19:32 CDT
====================================================================== 
Summary:                    "ast_bridge_call: Bridge failed" after
ChannelRedirect
Description: 
I want to create a feature which when activated by a person during a
conversation, transfers peer to some internal service (using
ChannelRedirect) and hangs up the person who activated the feature. This
basically works but causes WARNING message on the Asterisk console which is
probably not a good thing.
====================================================================== 

---------------------------------------------------------------------- 
 dimas - 08-30-07 19:32  
---------------------------------------------------------------------- 
Well, I re-run my tests with you patch and found that the patch only closes
half of issue. When caller activates feature everything is Ok, but when
callee activates it, I still have the warning - please see console output
for the whole call below:

    -- Executing [131 at ael-default:1] Set("SIP/1001-0a0c0130",
"_DYNAMIC_FEATURES=btfeat") in new stack
    -- Executing [131 at ael-default:2] Dial("SIP/1001-0a0c0130",
"Local/1002 at ael-default") in new stack
    -- Called 1002 at ael-default
    -- Executing [1002 at ael-default:1]
Macro("Local/1002 at ael-default-cf6c,2", "stdexten|1002|SIP/1002&IAX2/1002")
in new stack
    -- Executing [s at macro-stdexten:1] Set("Local/1002 at ael-default-cf6c,2",
"ext=1002") in new stack
    -- Executing [s at macro-stdexten:2] Set("Local/1002 at ael-default-cf6c,2",
"dev=SIP/1002&IAX2/1002") in new stack
    -- Executing [s at macro-stdexten:3]
Dial("Local/1002 at ael-default-cf6c,2", "SIP/1002&IAX2/1002/1002|20|") in new
stack
    -- Called 1002
[Aug 31 04:35:10] WARNING[11788]: app_dial.c:1106 dial_exec_full: Unable
to create channel of type 'IAX2' (cause 3 - No route to destination)

<XXX> please ignore this warning, I just have both IAX+SIP for this user
and only SIP is registered at the moment

    -- SIP/1002-0a0cf518 is ringing
    -- Local/1002 at ael-default-cf6c,1 is ringing
    -- SIP/1002-0a0cf518 answered Local/1002 at ael-default-cf6c,2
    -- Local/1002 at ael-default-cf6c,1 stopped sounds
    -- Local/1002 at ael-default-cf6c,1 answered SIP/1001-0a0c0130
  == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Local/1002 at ael-default-cf6c,2' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Local/1002 at ael-default-cf6c,2'
    -- Packet2Packet bridging SIP/1001-0a0c0130 and SIP/1002-0a0cf518
    -- Packet2Packet bridging SIP/1001-0a0c0130 and SIP/1002-0a0cf518
    --  Feature Found: btfeat exten: btfeat
    -- Executing [s at macro-bridgetest:1] NoOp("SIP/1002-0a0cf518",
"SIP/1001-0a0c0130") in new stack
    -- Executing [s at macro-bridgetest:2]
ChannelRedirect("SIP/1002-0a0cf518", "SIP/1001-0a0c0130|btcontext|1|1") in
new stack
[Aug 31 04:35:17] WARNING[11785]: res_features.c:1461 ast_bridge_call:
Bridge failed on channels SIP/1001-0a0c0130 and SIP/1002-0a0cf518, res =
-1
  == Spawn extension (btcontext, 1, 0) exited non-zero on
'SIP/1001-0a0c0130'
    -- Executing [1 at btcontext:1] Answer("SIP/1001-0a0c0130", "") in new
stack
    -- Executing [1 at btcontext:2] Morsecode("SIP/1001-0a0c0130", "SOS") in
new stack
    -- Executing [1 at btcontext:3] Hangup("SIP/1001-0a0c0130", "") in new
stack
  == Spawn extension (btcontext, 1, 3) exited non-zero on
'SIP/1001-0a0c0130' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-30-07 19:32  dimas          Status                   closed => feedback  
08-30-07 19:32  dimas          Resolution               fixed => reopened   
08-30-07 19:32  dimas          Note Added: 0069761                          
======================================================================




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