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

[Riley Fowle (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226006#comment-226006 ] 

Riley Fowle commented on ASTERISK-13145:
----------------------------------------

Gareth thanks for all your work on this patch!

All - I'm still fairly new at Linux & Asterisk so it took me several weeks of reading forums, docs & finally a rebuild to get everything ALMOST working. I'm using Cisco 9971's but for some reason the conference button only works the first time, then the next time I get the "Unable to complete conference" error on the phone after I press the Conference softkey to complete the conference.

I did a recent fresh install of the FreePBX distro which after updates had me running on Asterisk version 11.17.1. I followed Gareth's instructions to apply the patch using the same version as the patch, so after patch loaded correctly I'm now on 11.17.0.

Currently on firmware: sip9971.9-4-1-9

Have also tried these:

sip9971.9-4-2SR1-2
sip9971.9-3-2SR1-1
sip9971.9-3-1-33


In the GUI under Asterisk Sip Settings - Chan SIP:
NAT is set to Never
Under other SIP settings - 
tcpenable = yes
transport = tcp (currently have tcp, udp as it seemed to make no difference and we have a couple older phones I haven't upgraded yet)
subscribe = 3610 (have a line of these for each extension)
cisco_usecallmanager = yes (added this per a forum post but not sure it matters)


Under the extensions:

Nat: never (no)
Qualify: yes (tried it as no per some posts but had problems with dial plan working & didn't seem to help conference call issue)
Transport: TCP Only
Everything else is defaults I believe


In my SEPxxxxxxxx.cnf.xml file I built using the working example in the forums plus a few tweaks per Gareth's documentation. Here a few of the relevant settings to my issue as I understand it:

<featurePolicyFile>DefaultFP.xml</featurePolicyFile>  (This is Gareth's file renamed)
<transportLayerProtocol>4</transportLayerProtocol>
All my <sipLines> have: <proxy>USECALLMANAGER</proxy>

Can anyone offer any insight on my issue here? I am out of ideas so I appreciate any advice!

Thanks!


> [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
>            Assignee: Gareth Palmer
>         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-11.17.0.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, 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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