[asterisk-bugs] [Asterisk 0011753]: app_channelredirect relies on ast_parseable_goto which fails to redirect channels
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Jan 30 19:57:28 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11753
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Reported By: johan
Assigned To:
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Project: Asterisk
Issue ID: 11753
Category: Applications/app_channelredirect
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 98558
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 01-12-2008 16:43 CST
Last Modified: 01-30-2008 19:57 CST
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Summary: app_channelredirect relies on ast_parseable_goto
which fails to redirect channels
Description:
It seems like ChannelRedirect isn't working very well in trunk. I patched
ChannelRedirect to report the status in bug 0011553 for asterisk
1.4-trunk.
When I was porting this patch to trunk I stumbled on this issue.
The source av ast_parseable_goto that seems to fail even if there is a
vaild channel and destination. I have tested the following scenarious:
I place call http://bugs.digium.com/view.php?id=1 in either MusicOnHold(),
Meetme(), Playback() then I've a
call http://bugs.digium.com/view.php?id=2 that makes a
Channelredirect(call-numer-1-channelname,newcontext,newexten,1)
This always fails.
However if you do a core show channels after this unsucessful redirect you
will se:
Channel Location State Application(Data)
Zap/pseudo-598993578 s at default:1 Rsrvd (None)
SIP/callnumer1-08222 s at newcontext:0 Up MeetMe(1,dm)
Note the newcontext:0...
And in the case you redirect channel http://bugs.digium.com/view.php?id=1 where
it does a Playback() the
redirect will occur after the Playback is finished. This will not happen
with the other applications thou.
Maybe I'm making a misstake, but this confuses me a lot...
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johan - 01-30-08 19:57
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I've tried this once again hoping I was mistaken.
I checked out revision SVN-trunk-r101344
and tried the following extensions.ael
context catchall {
_X. => {
Answer();
Wait(1);
Wait(888);
}
}
After having one call waiting in Wait(888) I added the following row
before Wait(888):
ChannelRedirect(SIP/0303350971-082294c8,test,s,1);
Then I called in on another phone.
The output was the following:
-- Executing [0303350971 at catchall:1] Answer("SIP/0303350971-082294c8",
"") in new stack
-- Executing [0303350971 at catchall:2] Wait("SIP/0303350971-082294c8",
"1") in new stack
-- Executing [0303350971 at catchall:3] Wait("SIP/0303350971-082294c8",
"888") in new stack
mio*CLI> ael reload
<snip>
mio*CLI> core show channels
Channel Location State Application(Data)
SIP/0303350971-08229 0303350971 at catchall: Up Wait(888)
1 active channel
1 active call
1 call processed
== Using SIP RTP CoS mark 5
-- Executing [0303350971 at catchall:1] Answer("SIP/0303350971-0822ea30",
"") in new stack
-- Executing [0303350971 at catchall:2] Wait("SIP/0303350971-0822ea30",
"1") in new stack
-- Executing [0303350971 at catchall:3]
ChannelRedirect("SIP/0303350971-0822ea30",
"SIP/0303350971-082294c8,test,s,1") in new stack
-- Executing [0303350971 at catchall:4] Wait("SIP/0303350971-0822ea30",
"888") in new stack
mio*CLI> core show channels
Channel Location State Application(Data)
SIP/0303350971-0822e 0303350971 at catchall: Up Wait(888)
SIP/0303350971-08229 s at test:0 Up Wait(888)
2 active channels
2 active calls
2 calls processed
mio*CLI> core show version
Asterisk SVN-trunk-r101344 built by root @ mio on a i686 running Linux on
2008-01-31 01:43:24 UTC
Issue History
Date Modified Username Field Change
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01-30-08 19:57 johan Note Added: 0081480
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