[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=204110#comment-204110 ] 

Stuart Lape commented on ASTERISK-13145:
----------------------------------------

@Sean - thanks for the advice, yes I have, no joy - the code I currently have is:

[from-internal-custom]

; Strip the x-cisco-serviceuri- prefix
exten => _[x]-cisco-serviceuri-.,1,Goto(${EXTEN:19},1)
; Strip the x-cisco-serviceuri- prefix for cfwdall
exten => _[x]--serviceuri-.,1,Goto(${EXTEN:22},1)
; Enable forwarding
exten => _cfwdall-.,1,Answer
exten => _cfwdall-.,n,Set(SIPPEER(${CHANNEL(peername)},callforward)=${EXTEN:8})
exten => _cfwdall-.,n,Hangup(normal_clearing)
; Disable forwarding
exten => _cfwdall-.,1,Answer
exten => _cfwdall-.,Set(SIPPEER(${CHANNEL(peername)},callforward)=)
exten => _cfwdall-.,Hangup(normal_clearing)
; Call Pickup
exten => _blfpickup-.,1,Answer
exten => _blfpickup-.,n,Dial(Local/**${EXTEN:10}@from-internal)
exten => _blfpickup-.,n,Hangup(normal_clearing)

I get the call cannot be completed as dialled with that.

I've tried your code instead and with that I get the circuits busy signal when I enter any number.  Using IncrediblePi on Raspberry Pi if that helps.
                
> [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-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, 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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