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

[Andy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234248#comment-234248 ] 

Andy edited comment on ASTERISK-13145 at 12/14/16 3:31 PM:
-----------------------------------------------------------

I have a CP-8841 and I'm having problems getting a couple of things to work:

1.Paging to the phone with SIPCiscoPage. If I try to page the 8841 from another phone I can see the notification on the 8841's screen that I'm paging it, but it there's no audio and it doesn't go off hook like the older models do with the microphone muted and speaker enabled. Doesn't seem to matter if I use multicast or unicast.

2.Multiple lines registered to the phone. If I try just one additional line, the phone does not register. If I take the extra line away, it registers OK.

Has anyone worked with this model before and know how to get these two things going? Would appreciate any help, thanks.




was (Author: twofourniner):
I have a CP-8841 and I'm having problems getting a couple of things to work:

1.Paging to the phone with SIPCiscoPage. If I try to page the from another phone 8841 I can see the notification on the 8841's screen that I'm paging it, but it there's no audio and it doesn't go off hook like the older models do with the microphone muted and speaker enabled. Doesn't seem to matter if I use multicast or unicast.

2.Multiple lines registered to the phone. If I try just one additional line, the phone does not register. If I take the extra line away, it registers OK.

Has anyone worked with this model before and know how to get these two things going? Would appreciate any help, 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: 00_READ_ME_FIRST.txt, cisco-usecallmanager-11.17.0.patch, cisco-usecallmanager-11.17.1.patch, cisco-usecallmanager-11.18.0.patch, cisco-usecallmanager-11.19.0.patch, cisco-usecallmanager-11.20.0.patch, cisco-usecallmanager-11.21.2.patch, cisco-usecallmanager-11.22.0.patch, cisco-usecallmanager-11.23.0.patch, cisco-usecallmanager-11.24.1.patch, cisco-usecallmanager-11.25.0.patch, cisco-usecallmanager-13.10.0.patch, cisco-usecallmanager-13.12.1.patch, cisco-usecallmanager-13.13.0.patch, dialtemplate.xml, featurepolicy.xml, SEP000000000000.cnf.xml, softkeys.xml
>
>
> This patch provides support for Cisco 6900, 7900, 8800 and 9900 series phones using the SIP firmware.
> Available features are: Busy Lamp Field, Off Hook Notification, Call Forward, Do Not Disturb, Huntgroup Login, Call Park (Notify and Monitor), Server-Side Ad-Hoc Conference, Conference List, Kick and Mute/Unmute, Multi-Admin Conference, Multiple Lines via Bulk Register, Immediate Divert, Call Recording, Restart or Reset via CLI, Call Pickup Notification, Call Back, Join Calls, Mallicious Call ID, Quality Reporting Tool and Fail-over/Fail-back.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at [http://usecallmanager.nz] to see the additional configuration options required for the phones to operate correctly.



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