[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=234152#comment-234152 ] 

Andy commented on ASTERISK-13145:
---------------------------------

I've discovered what is causing the segfault I mentioned above, but I don't know why it's causing the segfault.
The culprits are seemingly some bulk registration entries I have on certain peers in my sip.conf.
Example: register=8100

I removed these and I can again restart asterisk without having to reboot my phones first. I had remembered I could restart asterisk without any problems pretty recently, and it turns out it must have stopped working when I added those bulk registration entries to peers in sip.conf. If I didn't need multiple registrations, I could just go without it but unfortunately that is not an option in my case.

Maybe I'm not using them correctly? Here's an example of a sip peer I have configured:

[100]
	cisco_usecallmanager=yes
        cisco_keep_conference=no
        cisco_multiadmin_conference=yes
        dndbusy=yes
        huntgroup_default=yes
        type=friend
	context=phones
        defaultuser=100
        callerid="That Guy" <100>
        nat=no
	trustrpid=no
        sendrpid=rpid
        rpid_update=yes
        rpid_immediate=yes
        allowsubscribe=yes
        notifyhold=yes
        callcounter=yes
        videosupport=no
        disallow=all
        allow=ulaw
        allow=alaw
        allow=g729
	secret=1234
	host=dynamic
        mailbox=100 at vm
	subscribe=101
        subscribe=102
        subscribe=104
        register=8100
	register=501
	register=500
	Callgroup=1
        pickupgroup=1

Any help is appreciated. 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, backtrace-ak12616.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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