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

[Gareth Palmer (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225245#comment-225245 ] 

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

I have debugged this further and the phone's behaviour of only sending a REGISTER for the first line is normal. 

The feature is called bulk-registration and the phone automatically switches to this mode when using TCP and a new-enough version of CUCM/CME.

An experimental patch (gareth-11.16.0.bulkregister.patch) has been provided.

To specify the additional lines for a phone there is a new {{sip.conf}} peer option called {{register}}. It will automatically register that peer when the primary peer registers.

{code:title=sip.conf}
[301](cisco-7961)
type=friend
...
register => 302
register => 303

[302](cisco-7961)
...

[303](cisco-7961)
...
{code}

*Note:* Some sip.conf options for bulk-registered peers have no effect.

1. {{subscribe}}: Only subscriptions for the primary peer are made.
2. {{qualify}}: OPTIONS ping is only sent to the primary peer, all lines get updated with the ping-time and reachability state.
3. DND and HLog: Only apply to the primary peer (this is a limitation of the phone).

Also, a {{lineIndex}} attribute needs to be specified for each {{<line/>}} feature definition. It must be the 1-based nth-index count of the line feature.

{code:title=SEPMAC.cnf.xml}
  <line button="1" lineIndex="1">
    <featureID>9</featureID>
    ...
  </line>
  ...
  <!-- some non-line feature definitions -->
  ...
  <line button="6" lineIndex="2">
    <featureID>9</featureID>
    ...
  </line>
{code}

When using multiple lines, the 7961 I tested with required that {{<missedCallLoggingOption/>}} be specified in SEPMAC.cnf.xml otherwise it wouldn't log missed calls. Documentation has been added for that option.

> [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-11.16.0.bulkregister.patch, gareth-11.16.0.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.



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