[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=207739#comment-207739 ] 

Jayant edited comment on ASTERISK-13145 at 7/6/13 3:35 AM:
-----------------------------------------------------------

I seem to be in the same boat as Darren.

Asterisk 11.3 and 9971 cisco sip9971.9-2-4-19. Also tried sip9971.9-3-2SR1-1. (I have access to older firmware if needed)

Video and Presence don't seem to work together - If i set cisco_usecallmanager=yes then video doesn't work, else the presence features don't work. Also have the same problem of low video quality on the phones. I have tested by calling between two 9971 phones, with other Video enabled SIP phones, the Bria app for iPhone and Android...

When the video does work, it works only by me setting the phone on mute and back. I get the same functionality on another 11.3 instance that has not been patched.

Will keep looking for clues to what i may be missing and will post back if i find a way forward.
                
      was (Author: jayant):
    I seem to be the same boat as Darren.

Asterisk 11.3 and 9971 cisco sip9971.9-2-4-19. Also tried sip9971.9-3-2SR1-1. (I have access to older firmware if needed)

Video and Presence don't seem to work together - If i set cisco_usecallmanager=yes then video doesn't work, else the presence features don't work. Also have the same problem of low video quality on the phones. I have tested by calling between two 9971 phones, with other Video enabled SIP phones, the Bria app for iPhone and Android...

When the video does work, it works only by me setting the phone on mute and back. I get the same functionality on another 11.3 instance that has not been patched.

Will keep looking for clues to what i may be missing and will post back if i find a way forward.
                  
> [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, 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.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira