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

Lincoln King-Cliby (JIRA) noreply at issues.asterisk.org
Tue Jan 14 12:37:09 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214026#comment-214026 ] 

Lincoln King-Cliby commented on ASTERISK-13145:
-----------------------------------------------

First -- much thanks goes to Gareth and the other contributors. We're running an armada of 796x on Asterisk 11.6.0 with the patch, and the features/functionality unleashed vs. stock asterisk is amazing. 

One issue I have noticed however, concerns voicemail indication on sets with two line appearances/directory numbers and 2 voicemail boxes. With stock asterisk, the behavior is what I would expect - e.g. a voicemail for "Line 1" shows the envelope for "Line 1", and a voicemail for "Line 2" shows the envelope for "Line 2". 

What we're seeing with the patch is a voicemail for "Line 2" will show the envelope/MWI for both lines 1 and 2, whole a voicemail for "Line 1" may show up on line 1, on both line 1 and 2, or not at all. Sometimes it will be "correct" for some period of time after a message is first left, but then become incorrect after. 

I'd be glad to provide any info that would help to troubleshoot this... I'm new enough to this "level" of Asterisk that I'm not even sure where I'd begin looking. 
                
> [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-11.2.1-dndbusy.patch, gareth-11.6.0.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, 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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