[asterisk-bugs] [JIRA] (ASTERISK-13145) [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML

Gareth Palmer (JIRA) noreply at issues.asterisk.org
Sun Sep 7 23:45:39 CDT 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222567#comment-222567 ] 

Gareth Palmer commented on ASTERISK-13145:
------------------------------------------

New Patch (gareth-11.12.0.patch) -

New Feature: New SIP global/peer option *huntgroup_default* \- sets the default hunt group login state of a peer.

New Feature: iDivert is now available for connected calls. Redirects call to *idivert* extension and sets *IDIVERT_PEERNAME* variable.

New Feature: CallBack is now supported. Phone can receive a notification when an extension becomes available.

To work correctly the following needs to be true \-

  \- CallBack must be either configured on a line-key, or CallBack soft-key must be included in On Hook and Ring Out states.

  \- There must be a hint for either the connected line number or the dialed number.

  \- A ring-tone called callback.raw must to be downloadable by the phone to get an audible notification.

Notification happens after the hint transitions from the INUSE/BUSY device state and/or from DND to AVAILABLE presence state.

So activating CallBack on a RINGING extension requires the hint to become INUSE/BUSY first. eg: phone goes off-hook and then on-hook.

New Feature: Indicate that the phone is call-forwarded to voice-mail if the target extension is *vmexten*.

New Feature: Include RTPTx stats in QRT report.

Documentation: Show which soft-keys are valid in which call-states.

Documentation: Add network locale information.

Documentation: Add user locale information.

Documentation: Add explanatory text to feature policy page.

> [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-13145
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-13145
>             Project: Asterisk
>          Issue Type: New Feature
>          Components: Channels/chan_sip/NewFeature
>            Reporter: David McNett
>         Attachments: 01-btias.patch, 02-media-attrib-sdp.patch, 03-media-attrib-sdp-backport.patch, 04-imageattr.patch, 7965.xml, 8001 to 8003 and hangup.pcap, 8001 to 8003 and pickup then talk then hangup.pcap, asterisk-1.8.7.0-chan_sip.patch, backtrace.txt, Capture - CSO Presence - Lift and Replace Handset.pcap, Capture - CSO Presence - Ring between 2 monitored extensions.pcap, chan_sip.c_available_on-the-phone.patch, chan_sip.c_blf_available_on-the-phone.patch, chan_sip.c.patch, cisco-blf-asterisk.1.6.0.26.patch, cisco-blf-asterisk.1.6.2.13.patch, cisco-blf-asterisk.1.8.0.patch, core-ast115-sccp.tar.gz, gareth-10.6.0.patch, gareth-11.12.0.patch, gareth-11.2.1-dndbusy.patch, gareth-1.8.14.0.patch, gareth-documentation-url.txt, gareth-featurepolicy.xml, gareth-mk-1.8.13.0.patch, gareth-softkeys.xml, gareth-softkeys.xml, memleak_astdb.patch, messages-1, Poly_reboot.log, rjw-11.4.0.patch, second-sip-trace-7941-9-1-1SR1.txt, sip-trace-7941-9-1-1SR1.txt, trace2.txt
>
>
> Cisco phones appear to be unable to parse the existing PIDF XML being generated by Asterisk for presence notification.  I've attached a patch which produces well-formed (but incomplete) XML which will satisfy a Cisco phone.  The patch as supplied will successfully render a "busy" subscription, but does not send a subsequent "available" notification, so presence detection only half works currently.
> I suspect the next step might be to watch some CallManager SIP traffic to identify precisely what XML tags the phone is expecting in order to properly parse an available subscription, but I'm not in a position to do that.  I'll continue to work with this, though, and perhaps may be able to stumble upon the precise data the Cisco phone is looking for.
> {{****** ADDITIONAL INFORMATION ******}}
> I believe that this requires the Cisco phones be configured to use SIP TCP when connecting to Asterisk.



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list