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

[Giovanni Di Cicco (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217637#comment-217637 ] 

Giovanni Di Cicco commented on ASTERISK-13145:
----------------------------------------------

HI all, apologies if I am asking a basic question but I am relatively new to Asterisk. I have read through this post a few times and have searched the internet but am still struggling.
I am running Asterisk 11.7.0 with Gareth Patch 11.7.0 & FreePBX 2.11.0.23.
I have a Cisco 9971 (ext 2001) , 7945 (ext 2002) and a  8945 (ext 2003). BLF is working for Busy & Ringing. Call Forwarding, BLF pick-up and video is also functioning well.
The issue I have is with the DND presence. It is still not clear to me to what is needed exactly for the DND presence to be reflected to the other phones. I used the SEP.cnf.xml example Gareth supplied on  26 June 2013 for the Cisco 9971 (extension 2001)

<line button="6">
    <featureID>130</featureID>
   <featureLabel>DND</featureLabel>
   <featureOptionMask>1</featureOptionMask>
</line>

The phone now sends a SIP “Publish” message when the button is toggled, but I am unable to see the PresenceState on the PBX:

pbx*CLI> core show hints
Location	Hints	DeviceState	PresenceState	Watchers
2003 at ext-local
SIP/2003,CustomPresence:2003	Idle		1
2002 at ext-local
SIP/2002,CustomPresence:2002	Idle		2
2001 at ext-local
SIP/2001,CustomPresence:2001	Idle		2


I have also toggled the DND by using “sip donotdisturb on/off 2001” but again the presence state is not reflected in the output of “core show hints”.

Am I missing some fundamental Presence config somewhere? 

Any help on this matter would be much appreciated.

@Gareth, the work you have done with this patch is fantastic, many thanks for sharing this with us! 


> [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-11.2.1-dndbusy.patch, gareth-11.7.0.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, 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.



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