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

Wed Oct 9 13:47:08 CDT 2013

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=210853#comment-210853 ] 

Eric Campbell commented on ASTERISK-13145:
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I am running the 11.4 version of the patch and having some specific issues with the 89xx line of Cisco phones. I have tried the 11.5 and 1.8.14 patches as well. I know that these have been worked on in the past and that they are supposed to work. 

My question specifically has to do with the phones and their reluctance to allow a missed call to leave the main display and get placed into the Missed calls log.

I am getting a retransmissions errors on unanswered calls. I have seen this behavior with phones both Local and remote (nat'ed). Just wondering if anyone has been able to get past this issue if they experienced it at all.

WARNING[19422] chan_sip.c: Retransmission timeout reached on transmission ccd539d7-0a17000a-39646475-5d2f4358 at 192.168.200.117 for seqno 102 (Critical Response)

I would also be interested in finding out if Floren would share the specs being used to successfully deploy these phones. Such as firmware, version of the patch and so forth.

Eric

> [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-11.2.1-dndbusy.patch, gareth-11.5.0.patch, gareth-1.8.14.0.patch, 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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