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Summary:ASTERISK-12320: [patch] Implementation of application/telephone-event for SIP INFO
Reporter:Alisher (licedey)Labels:
Date Opened:2008-07-05 11:20:26Date Closed:2011-06-07 14:03:01
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/NewFeature
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) chan_sip_svn1_4.diff
( 1) chan_sip.diff
Description:Currently Asterisk supports only application/dtmf-relay payload of SIP info messages. In the latest version application/dtmf (shortinfo) was added.
This patch implements support for telephone-event.
In order to set telephone-event dtmf mode for device set

dtmfmode=info_tel

in sip.conf

****** ADDITIONAL INFORMATION ******

http://www.ietf.org/rfc/rfc2833.txt
Comments:By: Olle Johansson (oej) 2008-07-05 13:17:57

All new functionality needs to be developed against trunk, not 1.4. Thanks.

By: Olle Johansson (oej) 2008-07-05 13:19:09

The RFC 2833 doesn't mention sending anything over SIP INFO. Who implements this and why?

By: Alisher (licedey) 2008-07-07 11:10:04

These ip phones implements it:
http://www.dasannetworks.com/english/solutions/html/solutions_main.asp?check=group

https://www.samsung070.com/images/business/phone/img_general_SMT-i3010.gif

Also there is a company Xener systems that supplies softswitches major telecom and ISP in Korea works supports it.
http://www.xener.com/servlet/DownloadSV?p_filepath=/test_board/Xener%20Introduction_(English)[1195816776446].pdf



By: Leif Madsen (lmadsen) 2008-12-05 08:29:14.000-0600

If someone wants to take this up and move it along, then please have a bug marshall on #asterisk-bugs on the IRC network irc.freenode.net have it reopened. However, since this is a feature request without a patch (with an accepted license) I'm going to suspend this for now.