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

Mon Jul 22 20:27:49 CDT 2019

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247636#comment-247636 ] 

Boris P. Korzun commented on ASTERISK-13145:
--------------------------------------------

There's a problem with Cisco 8945. It's look like a [#comment-246203]. 

I get a warning in the log before and after any call:
{noformat}
[Jul 23 10:22:43] WARNING[100472]: chan_sip.c:29301 int handle_notify_dialog(struct sip_pvt *, struct sip_request *): Unknown peer '2'
[Jul 23 10:22:46] WARNING[100472]: chan_sip.c:29301 int handle_notify_dialog(struct sip_pvt *, struct sip_request *): Unknown peer '2'
{noformat}
I've dumped the traffic and got the phone send _NOTIFY_ to unknown peer sip:2 at asterisk.domain.tld before and after any call:
BEFORE:
{noformat}
NOTIFY sip:2 at 192.168.8.11 SIP/2.0
Via: SIP/2.0/TCP 192.168.2.136:52928;branch=z9hG4bK25d1ff8b
To: "User-2870" <sip:2 at 192.168.8.11>
From: "User-2870" <sip:2 at 192.168.8.11>;tag=203a078206a200403ad37784-17d71883
Call-ID: 23e85239-622f44af at 192.168.2.136
Date: Fri, 26 Apr 2019 06:08:54 GMT
CSeq: 5 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:2870 at 192.168.2.136:52928;transport=tcp>
Authorization: Digest username="2870",realm="asterisk",uri="",response="ee1c64d881686a21e395a07dc0aec394",nonce="68d2ff18",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 501
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="4" state="partial" entity="sip:2870 at 192.168.2.136">
<dialog id="3" call-id="203a0782-06a20005-3c151ff4-20bf65c2 at 192.168.2.136" local-tag="203a078206a2003f1b8f4637-4d3b0964" remote-tag="" direction="initiator">
<state event="cancelled" code="0">trying</state>
<call:orientation>Unspecified</call:orientation>
<call:lock>unlocked</call:lock>
</dialog>
</dialog-info>
{noformat}

AFTER:
{noformat}
NOTIFY sip:2 at 192.168.8.11 SIP/2.0
Via: SIP/2.0/TCP 192.168.2.136:52928;branch=z9hG4bK7b304395
To: "User-2870" <sip:2 at 192.168.8.11>
From: "User-2870" <sip:2 at 192.168.8.11>;tag=203a078206a200413542d6cb-182608ee
Call-ID: 253a8bb6-18394804 at 192.168.2.136
Date: Fri, 26 Apr 2019 06:08:57 GMT
CSeq: 6 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:2870 at 192.168.2.136:52928;transport=tcp>
Authorization: Digest username="2870",realm="asterisk",uri="",response="03deaab27500895ad3c081454abe371c",nonce="61f91c71",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 505
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="5" state="partial" entity="sip:2870 at 192.168.2.136">
<dialog id="3" call-id="203a0782-06a20005-3c151ff4-20bf65c2 at 192.168.2.136" local-tag="203a078206a2003f1b8f4637-4d3b0964" remote-tag="" direction="initiator">
<state event="cancelled" code="0">terminated</state>
<call:orientation>Unspecified</call:orientation>
<call:lock>unlocked</call:lock>
</dialog>
</dialog-info>
{noformat}
where 192.168.2.136 - IP of the phone, 192.168.8.11 - IP of the Asterisk server

How to suppress the warning?

> [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
>              Labels: patch, pjsip
>         Attachments: 00_READ_ME_FIRST.txt, AppDialRules.xml, cisco-usecallmanager-13.25.0.patch, cisco-usecallmanager-13.26.0.patch, cisco-usecallmanager-13.27.0.patch, cisco-usecallmanager-16.2.0.patch, cisco-usecallmanager-16.3.0.patch, cisco-usecallmanager-16.4.0.patch, DialTemplate.xml, FeaturePolicy.xml, SEPMAC.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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