[asterisk-bugs] [Asterisk 0019190]: When calling myself with a SIP TRUNK with ITSP provider, incoming call is considered diverted and channel converted to Local/
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Apr 27 10:39:13 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=19190
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Reported By: albersag
Assigned To:
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Project: Asterisk
Issue ID: 19190
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.17.3
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: 2011-04-27 10:35 CDT
Last Modified: 2011-04-27 10:39 CDT
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Summary: When calling myself with a SIP TRUNK with ITSP
provider, incoming call is considered diverted and channel converted to Local/
Description:
Asterisk connected to SIP Provider, seems to bridge SIP Channels or Console
Channels as a call forwarding when they are incoming INVITES related to
outbound call made with same SIP provider account. ( I call myself).
When asterisk make this threatment, i could not use SIP functions s as
SIP_HEADER, because it is not a SIP CHANNEL as it has been considered a
call diverted.
I attach sip trace
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(0134183) albersag (reporter) - 2011-04-27 10:39
https://issues.asterisk.org/view.php?id=19190#c134183
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Im using this simple incoming dialplan
[from-voztele]
exten => s,1,Noop(${SIP_HEADER(To)})
exten => s,n,Set(dialed=${SIP_HEADER(To):5:9})
exten => s,n,Goto(ext-did,${dialed},1)
But for incoming call, my own DIDs is not usable, as SIP_HEADER is not
usable because channel was converted to Local/
Issue History
Date Modified Username Field Change
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2011-04-27 10:39 albersag Note Added: 0134183
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