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

[Stuart Lape (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222746#comment-222746 ] 

Stuart Lape edited comment on ASTERISK-13145 at 9/23/14 2:05 PM:
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@Jayant I've been trying pretty extensively with that code and absolutely no joy.  There's a hangup almost instantly as I initiate it, here's my CLI output:

    -- Executing [record at from-internal:3] MixMonitor("SIP/202-00000003", "/var/spool/asterisk/record/1411498391.0-202-in.wav,,,akqx") in new stack
    -- Executing [record at from-internal:4] Hangup("SIP/202-00000003", "normal_clearing") in new stack
  == Spawn extension (from-internal, record, 4) exited non-zero on 'SIP/202-00000003'
    -- Executing [h at from-internal:1] Hangup("SIP/202-00000003", "") in new stack


was (Author: laptopu):
@Jayant I've been trying pretty extensively with that code and absolutely no joy.  There's a hangup almost instantly as I initiate it, here's my CLI output:

[code]
    -- Executing [record at from-internal:3] MixMonitor("SIP/202-00000003", "/var/spool/asterisk/record/1411498391.0-202-in.wav,,,akqx") in new stack
    -- Executing [record at from-internal:4] Hangup("SIP/202-00000003", "normal_clearing") in new stack
  == Spawn extension (from-internal, record, 4) exited non-zero on 'SIP/202-00000003'
    -- Executing [h at from-internal:1] Hangup("SIP/202-00000003", "") in new stack
[/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-11.12.0.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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