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

[Andrew (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Andrew updated ASTERISK-13145:
------------------------------

    Comment: was deleted

(was: Hi All,

Having issues with getting the new presence state support working in the v11 patches and Cisco 7975 phones. All phones are not showing any presence support at all.

I have modified the hints to be duplicated as per previous comment by Gareth (i.e. exten => 6006,hint,SIP/6006,SIP/6006) but see below how device state is showing inuse for 6003, PresenceState still shows as available for some reason?


{noformat}onslowbox*CLI> core show hints
Location                  Hints                           DeviceState     PresenceState   Watchers
6003 at local_sip_phone      SIP/6003,SIP/6003               InUse           available       8 
6002 at local_sip_phone      SIP/6002,SIP/6002               Idle            available       4 
6001 at local_sip_phone      SIP/6001,SIP/6001               Idle            available       6 
6007 at local_sip_phone      SIP/6007,SIP/6007               Idle            available       7 
6006 at local_sip_phone      SIP/6006,SIP/6006               Idle            available       7 
6005 at local_sip_phone      SIP/6005,SIP/6005               Unavailable                     3 
6012 at local_sip_phone      SIP/6012,SIP/6012               Idle            available       3 
6017 at local_sip_phone      SIP/6017,SIP/6017               Idle            available       7 
6014 at local_sip_phone      SIP/6014,SIP/6014               Unavailable                     4 
6060 at local_sip_phone      SIP/6060,SIP/6060               Idle            available       7 
{noformat}

Any suggestions?)
    
> [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, gareth-10.6.0.patch, gareth-11.2.1-dndbusy.patch, gareth-11.2.1.patch, gareth-1.8.14.0.patch, gareth-featurepolicy.xml, gareth-mk-1.8.13.0.patch, gareth-softkeys.xml, gareth-softkeys.xml, memleak_astdb.patch, messages-1, Poly_reboot.log, 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.

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