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

Mon Dec 3 12:54:45 CST 2012

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=200340#comment-200340 ] 

BastianM edited comment on ASTERISK-13145 at 12/3/12 12:53 PM:
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@Gareth

Thank you very much for your help. I've pretty much managed to resolve both issues. The key thing here is that presence state seems to be case sensitive. Attached is my working (11.0.1) extensions_custom.conf file: http://pastebin.com/3JyiZUte

I'm getting an error message but it doesn't seem to affect functionality:

[2012-12-03 19:51:29] NOTICE[1760] func_presencestate.c: Not doing any base64 decoding

I've also managed to get video to work, albeit with an undesired side effect, which is that in order to see video on the 9971 I have to mute and then unmute the video. The log shows: "chan_sip.c: Ignoring video stream offer because port number is zero"

      was (Author: bastian):
    @Gareth

Thank you very much for your help. I've pretty much managed to resolve both issues. The key thing here is that presence state seems to be case sensitive. Attached is my working (11.0.1) extensions_custom.conf file: http://pastebin.com/3JyiZUte

I've also managed to get video to work, albeit with an undesired side effect, which is that in order to see video on the 9971 I have to mute and then unmute the video. The log shows: "chan_sip.c: Ignoring video stream offer because port number is zero"

> [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-11.0.1.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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