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

Wed Oct 16 23:24:14 CDT 2013

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=210985#comment-210985 ] 

Floren Munteanu edited comment on ASTERISK-13145 at 10/16/13 11:23 PM:
-----------------------------------------------------------------------

Hi Eric,

I'm currently running Asterisk 11.5.1 with Gareth's patch.
The firmware I installed on my 8961's is sip8961.9-3-4-24, everything works as expected. The retransmission issues previously posted on #comment-207989 don't affect the phone functionality. After spending some time on the #asterisk IRC channel, some of the experts posted that the retransmissions are part of the Cisco's "wacky crazy" SIP comm.

I would really like to have Gareth take a look at the retransmissions and hopefully explain what they actually do. I think the most important thing for troubleshooting is to access the phone actual log and see if there are any cnf.xml configurations errors. I know I did found few on my own, even after weeks of extensive research done on the Internet. Here it is how to check the phone logs:

{noformat}
[floren at ZEUS ~]$ ssh -l cisco cisco
cisco at cisco's password: [see your cnf.xml]
(none) login: default
Password: cisco
Welcome to MontaVista(R) Linux(R) Professional Edition Blackfoot (0702518).
Cisco IP Phone 8961 9-3-4-24
$ /usr/sbin/debugsh
DEBUG> debug sip-adapter ccapp sip-state sip-messages sip-reg-state gsm lsm fsm ccapp remote-cc
DEBUG> debug jvm Config debug
DEBUG> debug jvm SESSIONMANAGER debug
DEBUG> debug jvm Properties debug
DEBUG> debug jvm callhist debug
DEBUG> quit
$ /usr/sbin/sdump
{noformat}

      was (Author: teck):
    Hi Eric,

I'm currently running Asterisk 11.5.1 with Gareth's patch.
The firmware I installed on my 8961's is sip8961.9-3-4-24, everything works as expected. The retransmission issues previously posted on #comment-207989 don't affect the phone functionality. After spending some time on the #asterisk IRC channel, some of the experts posted that the retransmissions are part of the Cisco's "wacky crazy" SIP comm.

I would really like to have Gareth take a look at the retransmissions and hopefully explain what they actually do. I think the most important thing for troubleshooting is to access the phone actual log and see if there are any cnf.xml configurations errors. I know I did found few on my own, even after weeks of extensive research done on the Internet. Here it is how to check the phone logs:

{noformat}
$ ssh -l cisco cisco
cisco at cisco's password: [see your cnf.xml]
(none) login: default
Password: cisco
Welcome to MontaVista(R) Linux(R) Professional Edition Blackfoot (0702518).
Cisco IP Phone 8961 9-3-4-24
$ /usr/sbin/debugsh
DEBUG> debug sip-adapter ccapp sip-state sip-messages sip-reg-state gsm lsm fsm ccapp remote-cc
DEBUG> debug jvm Config debug
DEBUG> debug jvm SESSIONMANAGER debug
DEBUG> debug jvm Properties debug
DEBUG> debug jvm callhist debug
DEBUG> quit
$ /usr/sbin/sdump
{noformat}

> [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.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira