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

[Jayant (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217649#comment-217649 ] 

Jayant commented on ASTERISK-13145:
-----------------------------------

@Giovanni:

Here's a very helpful article that I used first to understand how best to use this patch: [http://www.freepbx.org/forum/freepbx/tips-and-tricks/cisco-9971-phone-configuration-working-example-with-setup-tips] You can follow that approach to setup a custom DND button or you can follow a slightly different approach below.

I'm guessing you have a problem in the way you've defined hints. This is what I did to get DND status reflected:

Add this in extensions_override_freepbx.conf if you use Freepbx. Else it can go directly into the dialplan right where you define the extension. (7101,7102,7106 are extension numbers - replace for the numbers on your dialplan)
{code}[ext-local]
exten => 7101,hint,SIP/${EXTEN},SIP/${EXTEN}
exten => 7102,hint,SIP/${EXTEN},SIP/${EXTEN}
exten => 7106,hint,SIP/${EXTEN},SIP/${EXTEN}{code}

This defines the appropriate hints that will be updated by the DND button. If you've subscribed to the hints correctly as described elsewhere on this post then you should start to see other phones reflect DND status correctly.

Here's what I get on the CLI with this approach:
{code}napbx2*CLI> sip donotdisturb on 7101
Do Not Disturb on '7101' enabled
napbx2*CLI> core show hints
Location                  Hints                           DeviceState     PresenceState   Watchers
7102 at ext-local            SIP/7102,SIP/7102               Idle                            1 
7101 at ext-local            SIP/7101,SIP/7101               Idle            DND             1 
7106 at ext-local            SIP/7106,SIP/7106               Unavailable                     0 
{code}

> [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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