[asterisk-bugs] [Asterisk 0012036]: [patch] RFC 3372 SIP-T receive implementation
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Feb 29 11:03:02 CST 2008
The following issue has been UPDATED.
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http://bugs.digium.com/view.php?id=12036
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Reported By: gasparz
Assigned To:
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Project: Asterisk
Issue ID: 12036
Category: Channels/NewFeature
Reproducibility: always
Severity: feature
Priority: normal
Status: new
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 02-20-2008 07:47 CST
Last Modified: 02-29-2008 11:03 CST
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Summary: [patch] RFC 3372 SIP-T receive implementation
Description:
The patch enables you to receive sip-t packets and decodes some fields.
The sip-t means that in the initial invite the content is multipart with
1) the sdp part
2) application/isup part
this second part contains the IAM message from the SS7. This is cool
because it contains some fields that are not available in the basic sip
packet like:
Origination line:
0- Plain Old Telephone Service (POTS) - non-coin service requiring no
special treatment
70- Code 70 identifies a line connected to a pay station (including both
coin and coinless stations) which does not use network provided coin
control signaling. II 70 is used to identify this type pay station line
irrespective of whether the pay station is provided by a LEC or a non-LEC.
II 70 is transmitted from the originating end office on all calls made from
these lines.
For some carriers this is the method to send the payphone information. If
you use toll free numbers this is a cool feature to have, so that you can
bill the payphone too.
Let me know how it worked for you. (For me it's working well).
Cheers
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Issue History
Date Modified Username Field Change
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02-29-08 11:03 file Category
Channels/chan_sip/General => Channels/NewFeature
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