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

Thu Mar 31 11:00:01 CDT 2016

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230099#comment-230099 ] 

Nick Montgomery commented on ASTERISK-13145:
--------------------------------------------

Hi there,

I'm having a bit of an issue on a system that I can't seem to figure out. I'm taking advantage of the SIPCiscoPage application introduced in release 11.20.0 of the patch.
I am running Asterisk 11.21.2 with Incredible GUI 12.0.30. The PBX is setup with an ATA for overhead paging in conjunction with paging all handsets which are Cisco 7961s.
To do this, I have setup a paging group in the FreePBX GUI that pages the ATA's extension as well as a virtual extension that dials the SIPCiscoPage application like so: local/802 at from-internal

This is the dialplan I am using for paging the handsets:

exten => 802,1,Set(GROUP()=paging)
same => next,ExecIf($[${GROUP_COUNT(paging)} != 1]?Hangup(user_busy))
;Unicast audio
same => next,SIPCiscoPage(102&103&105&107&108&110&111&112&200,v(40)d(From: ${CALLERID(number)}))
same => next,Hangup(normal_clearing)

This works great on the system we have here at our office, but at another location we are seeing the calls to the handsets start to fail after a period of time and they will no longer receive pages; however, they can still place and receive calls. This is fixed by rebooting the handsets.

This is the Asterisk output when the SIPCiscoPage app is executed.

-- Executing [802 at from-internal:3] SIPCiscoPage("Local/802 at from-internal-00000b72;2", "102&103&105&107&108&110&111&112&200,v(40)d(From: 100)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Local/802 at from-internal-00000b72;1 answered Local/PAGE801 at app-paging-00000b71;2
    -- <Local/802 at from-internal-00000b72;1> Playing 'beep.gsm' (language 'en')
[2016-03-31 10:43:21] NOTICE [13834]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:3e26fce6 failed, call miserably fails.
[2016-03-31 10:43:21] NOTICE [4209]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:192dac08 failed, call miserably fails.
[2016-03-31 10:43:21] NOTICE [5087]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:436d380a failed, call miserably fails.
    -- Executing [800 at app-pagegroups:14] ConfBridge("SIP/100-00000e1a", "1459439001263,,,admin_menu") in new stack
    -- SIP/8200-00000e1b is ringing
[2016-03-31 10:43:22] NOTICE[13834]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:39a5784c failed, call miserably fails.
[2016-03-31 10:43:22] NOTICE[4209]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:047984a5 failed, call miserably fails.
[2016-03-31 10:43:22] NOTICE[5087]: chan_sip.c:24627 handle_response_refer: SIP transfer to cid:18c2c847 failed, call miserably fails.
    -- SIP/8200-00000e1b answered Local/PAGE8200 at app-paging-00000b70;2
    -- <SIP/8200-00000e1b> Playing 'beep.gsm' (language 'en')

I have not been able to reproduce this problem on PBXs with similar configurations and I have not been able to observe when the problem starts.

Any insight on this or at least where I might look to find a solution would be very much 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, 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, 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, Call Back, Join Calls, Mallicious Call ID and Quality Reporting Tool.
> 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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