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

[Adam Horn (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233450#comment-233450 ] 

Adam Horn commented on ASTERISK-13145:
--------------------------------------

One phone that is reporting its status correctly gives this when hang up happens:
{code}
<--- SIP read from TCP:xx.xx.xx.xx:54112 --->
NOTIFY sip:1001 at xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/TCP xx.xx.xx.xx:51104;branch=z9hG4bK1424ee88
To: <sip:1001 at xx.xx.xx.xx>
From: <sip:1001 at xx.xx.xx.xx>;tag=aca016fd5e7b001a8db5e7a0-d2440cd6
Call-ID: 984389b8-8315b3ae at xx.xx.xx.xx
Date: Mon, 07 Nov 2016 14:06:43 GMT
CSeq: 8 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:1001 at xx.xx.xx.xx:51104;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 363
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="7" state="partial" entity="sip:1001 at xx.xx.xx.xx">
<dialog id="8" call-id="aca016fd-5e7b0008-534066f8-f865496e at xx.xx.xx.xx" local-tag="aca016fd5e7b0018730a3790-a6480926"><state>terminated</state></dialog>
</dialog-info>
<------------->
{code}

One that doesn't report its presence correctly, this:
{code}
<--- SIP read from TCP:xx.xx.xx.xx:52812 --->
NOTIFY sip:1002 at xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/TCP xx.xx.xx.xx:49740;branch=z9hG4bK80c26665
To: <sip:1002 at xx.xx.xx.xx>
From: <sip:1002 at xx.xx.xx.xx>;tag=0022905b9a160344d8ce8f7b-c23afec6
Call-ID: 4dedb331-54e0b064 at xx.xx.xx.xx
Date: Mon, 07 Nov 2016 14:01:53 GMT
CSeq: 138 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:1002 at xx.xx.xx.xx:49740;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 367
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="137" state="partial" entity="sip:1002 at xx.xx.xx.xx">
<dialog id="165" call-id="0022905b-9a160054-daa9e75f-b30d2c5c at xx.xx.xx.xx" local-tag="0022905b9a1603424532f52d-c0b15996"><state>terminated</state></dialog>
</dialog-info>
<------------->
{code}

> [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-13.10.0.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://docs.acsdata.co.nz/asterisk-cisco to see the additional configuration options required for the phones to operate correctly.



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