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

Stuart Lape (JIRA) noreply at issues.asterisk.org
Wed Sep 24 16:41:39 CDT 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222763#comment-222763 ] 

Stuart Lape commented on ASTERISK-13145:
----------------------------------------

I agree this should be adopted in Asterisk, however I don't have the skills to do this unfortunately.

I think if this was submitted it's fairly likely to be adopted, then we'll possibly get more support from the wider community as I bet only a small proportion of Cisco users are even aware of this patch.

Going back to my recording issue I'm still desperately trying to get call recording down to one mixed audio file instead of two channels.  In the MixMonitor suggestion a couple of posts up, is that definitely the right way to phrase the line in extensions.conf?  (or extensions_custom.conf for me as I'm on PIAF/FreePBX).  Monitoring the CLI the line just hangs up the instant call recording is initiated if I have the following:

; Enable recording
exten => record,1,Answer
same => next,Wait(1)
same => next,MixMonitor(${ASTSPOOLDIR}/record/${RECORD_UNIQUEID}-${RECORD_PEERNAME}-${RECORD_DIRECTION}.wav,,,akqx);
same => next,Hangup(normal_clearing)

And if I do the following I get a successful recording but as two files which I don't want:

; Enable recording
exten => record,1,Answer
same => next,Wait(1)
same => next,Record(${ASTSPOOLDIR}/record/${RECORD_UNIQUEID}-${RECORD_PEERNAME}-${RECORD_DIRECTION}.wav,,,akqx);
same => next,Hangup(normal_clearing)

Thanks again all for any help you can offer on this.

> [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
>         Attachments: 01-btias.patch, 02-media-attrib-sdp.patch, 03-media-attrib-sdp-backport.patch, 04-imageattr.patch, 7965.xml, 8001 to 8003 and hangup.pcap, 8001 to 8003 and pickup then talk then hangup.pcap, asterisk-1.8.7.0-chan_sip.patch, backtrace.txt, Capture - CSO Presence - Lift and Replace Handset.pcap, Capture - CSO Presence - Ring between 2 monitored extensions.pcap, chan_sip.c_available_on-the-phone.patch, chan_sip.c_blf_available_on-the-phone.patch, chan_sip.c.patch, cisco-blf-asterisk.1.6.0.26.patch, cisco-blf-asterisk.1.6.2.13.patch, cisco-blf-asterisk.1.8.0.patch, core-ast115-sccp.tar.gz, gareth-10.6.0.patch, gareth-11.12.0.patch, gareth-11.2.1-dndbusy.patch, gareth-1.8.14.0.patch, gareth-documentation-url.txt, gareth-featurepolicy.xml, gareth-mk-1.8.13.0.patch, gareth-softkeys.xml, gareth-softkeys.xml, memleak_astdb.patch, messages-1, Poly_reboot.log, rjw-11.4.0.patch, second-sip-trace-7941-9-1-1SR1.txt, sip-trace-7941-9-1-1SR1.txt, trace2.txt
>
>
> Cisco phones appear to be unable to parse the existing PIDF XML being generated by Asterisk for presence notification.  I've attached a patch which produces well-formed (but incomplete) XML which will satisfy a Cisco phone.  The patch as supplied will successfully render a "busy" subscription, but does not send a subsequent "available" notification, so presence detection only half works currently.
> I suspect the next step might be to watch some CallManager SIP traffic to identify precisely what XML tags the phone is expecting in order to properly parse an available subscription, but I'm not in a position to do that.  I'll continue to work with this, though, and perhaps may be able to stumble upon the precise data the Cisco phone is looking for.
> {{****** ADDITIONAL INFORMATION ******}}
> I believe that this requires the Cisco phones be configured to use SIP TCP when connecting to Asterisk.



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