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

[Lincoln King-Cliby (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=254296#comment-254296 ] 

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

Hi Gareth and team thanks for the excellent work on this patch. 

Have an odd issue that I'm hoping someone smarter than me can point in the right direction: 

We've been using 7961s for around a decade now; over the past year we "de-officed" and moved Asterisk to one of our colocation facilities. Everyone has a site-to-site VPN between their home offices and the colo facility. Currently running Asterisk 16.16.0 on Ubuntu 16.04.7 LTS

We've decided to selectively upgrade a few phones to 8861/8865s. I've had an 8865 (running 12.8(1)SR1) on my desk for a couple weeks now with no problems worth noting. One of my colleagues started using his 8865 yesterday and has complained that with some regularity it will display "Phone is Registering". Today I was on an (audio only) call with him and he reported that the phone banner changed to "Server Connection Lost" and the active line key was displaying "Call Preservation Mode", and I noticed his speed dial key on my phone had turned black. He couldn't place me on hold, start a new call, etc. but our call continued normally.

As soon as we hung up, his phone did the "Phone is Registering" cycle and returned to normal. I happened to have the Asterisk CLI up at the time and didn't notice anything odd pop out in console. 

Looking at the phone logs it appears that about every 4 hours a "LastTimeCMresetTCP" is getting logged :
[7:53:12am 23/03/21] LastTimeCMresetTCP
[...]
[11:53:18am 23/03/21] LastTimeCMresetTCP

Which would seem to correlated with the "registering" cycles. Googling that isn't yielding anything super useful. Looking at the VPN stats, switchport details, etc. outside of Asterisk everything looks normal -- and my phone with virtually the same config (save for speed dial keys and of course line appearances) isn't doing this. 

Any idea where I should start to look? 

> [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: Gareth Palmer
>            Assignee: Gareth Palmer
>            Severity: Major
>              Labels: patch, pjsip
>         Attachments: 00_READ_ME_FIRST.txt, AppDialRules.xml, cisco-usecallmanager-13.38.2.patch, cisco-usecallmanager-16.16.2.patch, cisco-usecallmanager-18.2.2.patch, DialTemplate.xml, FeaturePolicy.xml, SEPMAC.cnf.xml, SoftKeys.xml, usecallmanager-port.sh
>
>
> 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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)