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

[kuj (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=206028#comment-206028 ] 

kuj commented on ASTERISK-13145:
--------------------------------

Great work, Gareth! And thanks for not giving up on this. I wish these enhancements were becoming part of the release, after more than 4 years of you working on them! Thanks for staying on it!

Just installed 11.3.0-patch (on a Raspberry Pi, no less), and it works very well. Just two minor things I've observed:

1. In ConfLst, while the Mute button works fine, I get no visual indication whether a party is muted or not. On my 7970, all I see is a speaker icon and the caller ID name of the participant. Muting/unmuting doesn't change any of that, nor does pressing "Refresh"

2. For the BLF notification, would it not be more intuitive to assume the same subscribe context that the line/peer is in, instead of the "default" context? IOW, if the phone/peer registers in a non-default (e.g. "customer1") context, wouldn't defaulting any subscriptions to the @customer1 context be more intuitive? If one wanted to subscribe to extensions in a different context ("default" or others), one could always override in sip.conf.

Thanks for your great work!
                
> [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-11.3.0.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, 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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