[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=222770#comment-222770 ] 

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

@Stuart,

Can you try increasing the log level on the console to capture the reason for the call drops? Is it possible that there is a permissions problem or a path problem somewhere? Does this folder exist on your asterisk box - /var/spool/asterisk/record/ 

I presume the two files are getting created in the folder above with the record command. If so, read on.

I'm still running version 11.7 of the patch on the box where I had used the recordings feature - not sure if something has changed in the current version of the patch. That asterisk machine is still in use but I have not used recordings for a while now. I will try and find the time in the next 2-3 days to look into all my settings again and report back if I find something useful. 

> [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-11.12.0.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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