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

[Mat Phillips (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230605#comment-230605 ] 

Mat Phillips commented on ASTERISK-13145:
-----------------------------------------

Tried that but it's problematic (though I'll test again). What I found would happen is that the phone with the lower callbitrate setting would use it, and send the smaller amount of traffic over the bad link, but the extension it was calling would send the bitrate it was told to, the higher one. So only one device would 'play nice' with the congested link, the other phone just tried to shove the HQ bitstream down it regardless.

This may be because I'm not doing a sip reinvite, so each phone is talking to the asterisk server and the phone on the HQ side is quite happy to send that traffic and seems not to notice that the phone on the other leg cannot handle it.

We're planning to upgrade the connection anyway so it will be moot at some point.

> [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, 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, Call Back, Join Calls, Mallicious Call ID and Quality Reporting Tool.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at http://docs.acsdata.co.nz/asterisk-cisco to see the additional configuration options required for the phones to operate correctly.



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