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

[Andy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234267#comment-234267 ] 

Andy commented on ASTERISK-13145:
---------------------------------

Multiple lines are possible on the 8841 which I'm sure others already knew, however it is a different method than what I've used with some older phones I have.

I had to edit the SEPMAC.cnf.xml and add a line and line index section like I might in any other situation, however I removed the "USECALLMANAGER" from the <proxy> section in the line, as well as the authname and password. In sip.conf I have a register line for whatever additional extension or extensions I want to add, as outlined by the documentation on usecallmanager.nz.

This allows Asterisk to register the additional extensions and the phone does show the lines as available, however no outgoing calls can be made from them. Incoming calls work fine. As also mentioned in the documentation, lines registered in this way cannot be subscribed to, so you can't see if that line happens to be ringing from another phone.

It seems the 7961,62,70, and 75 all support multiple registrations in the way that allows you to make calls on them(not just receive), and other phones can subscribe to the lines as well.

The question on my mind is why is it that 79xx's that have the available keys seem to be able to have "true" multiple registrations, and the 8841 cannot? They all seem to register in the same way, but somehow the 8841 gets balled up when it tries to register the next line after the primary one, believing that it is trying to again register the same line twice when it actually isn't.

The error that appears in Asterisk is this:

WARNING[11558] chan_sip.c: username mismatch, have <someextension>, digest has <someotherextension>

I'm not expecting anyone to magically come up with some fix for this, unless of course one exists. I just wanted to put what I've found out for others to see since there's not a whole lot out there about this issue. I found a couple of posts from pbx in a flash and free pbx that pretty much concluded with the same things I did.


> [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-11.24.1.patch, cisco-usecallmanager-11.25.0.patch, cisco-usecallmanager-13.10.0.patch, cisco-usecallmanager-13.12.1.patch, cisco-usecallmanager-13.13.0.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://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)