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

[BosHab J (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205032#comment-205032 ] 

BosHab J commented on ASTERISK-13145:
-------------------------------------

I am trying to get CFwdAll button to forward all calls to another extension. I primarily need this to work on a Cisco 7911 IP phone. I am trying Gareth latest 11.3.0 patch on a new PIAF-Green (PIAF-2.0.6.4.2-VirtualMachine) on a Virtual Box. Patching was clean and compiling was successful. The thing is, It kind of half works with the 7911 phone running SIP11.9-2-1 Firmware. So I tried the same with a Cisco 7975 phone running SIP75.9-2-1 Firmware and it worked like a charm. 
On the 7911 after I press cfwdall button I hear two beeps and it prompts me to enter extension. After I do that nothing happens. But on asterisk I issue "sip show peer xxxx" showing the callforward used as required. However, the 7911 phone apparently does seem to like what ever it is that asterisk is doing. The reason I say that is because when I press the CFwdAll a second time to cancel it, the phone is prompting me again to enter extension number! Also it never shows on the screen that "call forwarding" is enabled like it did on the 7975. I am not sure if this patch is supposed to work with a 7911 or not. But I would like someone to help me sort this out.

Thank you for the wonderful work!


                
> [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, 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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