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

[Robin (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247658#comment-247658 ] 

Robin commented on ASTERISK-13145:
----------------------------------

Hi guys,

I am having a strange issue with my Cisco 8945 phone, loaded with latest SIP 9.4.2 firmware. The phone keeps showing its Caller ID as Anonymous in the INVITE request, though I have disabled it with *<callerIdBlocking>0</callerIdBlocking>*. I'm using *UDP Transport*.

A sample capture for the INVITE packet:

{quote}
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:[CALLEE_EXT]@[SIP_SERVER];user=phone SIP/2.0
    Message Header
        Via: SIP/2.0/UDP [PHONE_IP]:5060;branch=z9hG4bK58106e15
        From: "Anonymous" <sip:Anonymous@[SIP_SERVER]>;tag=203a0783aecd0009224ea09a-376c7fae
            SIP Display info: "Anonymous"
            SIP from address: sip:Anonymous@[SIP_SERVER]
            SIP from tag: 203a0783aecd0009224ea09a-376c7fae
        To: <sip:[CALLEE_EXT]@[SIP_SERVER]>
            SIP to address: sip:[CALLEE_EXT]@[SIP_SERVER]
        Call-ID: 203a0783-aecd0004-56539248-7cd3edc8@[PHONE_IP]
        [Generated Call-ID: 203a0783-aecd0004-56539248-7cd3edc8@[PHONE_IP]]
        Max-Forwards: 70
        Date: Sat, 27 Jul 2019 15:42:00 GMT
        CSeq: 101 INVITE
        User-Agent: Cisco-CP8945/9.4.2
        Contact: <sip:[CALLER_EXT]@[PHONE_IP]:5060;transport=udp>
            Contact URI: sip:[CALLER_EXT]@[PHONE_IP]:5060;transport=udp
        Expires: 180
        Accept: application/sdp
        Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
        Remote-Party-ID: "[CALLER_EXT]" <sip:[CALLER_EXT]@[SIP_SERVER]>;party=calling;id-type=subscriber;privacy=full;screen=yes
{quote}

My SIP server however doesn't want caller to hide its ID, and reject the INVITE request above with *SIP/2.0 403 Spoofed From-URI detected*.

Partial SEPMAC.xml:

{quote}
<sipCallFeatures>
      <cnfJoinEnabled>true</cnfJoinEnabled>
      <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
      <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
      <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
      <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
      <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
      <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
      <rfc2543Hold>true</rfc2543Hold>
      <callHoldRingback>2</callHoldRingback>
      <localCfwdEnable>true</localCfwdEnable>
      <semiAttendedTransfer>true</semiAttendedTransfer>
      <anonymousCallBlock>2</anonymousCallBlock>
      <callerIdBlocking>0</callerIdBlocking>
      <dndControl>0</dndControl>
      <remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
{quote}

Any idea on how to let this phone shows its Caller ID?

Cheers

> [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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