[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 10:42:29 CDT 2011
The following issue has been UPDATED.
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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: closed
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:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2011-04-27 10:35 CDT
Last Modified: 2011-04-27 10:42 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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(0134184) pabelanger (administrator) - 2011-04-27 10:42
https://issues.asterisk.org/view.php?id=19190#c134184
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That is because you are using the console command, look into Originate().
This is not a bug, but a support request.
---
Thanks for your comments. This does not appear to be a bug report and we
are closing it. We appreciate the difficulties you are facing, but it would
make more sense to raise your question in the support tracker,
http://www.asterisk.org/support
Issue History
Date Modified Username Field Change
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2011-04-27 10:42 pabelanger Note Added: 0134184
2011-04-27 10:42 pabelanger Status new => closed
2011-04-27 10:42 pabelanger Resolution open => no change
required
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