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

Sat Oct 26 19:58:08 CDT 2013

    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211260#comment-211260 ] 

Floren Munteanu edited comment on ASTERISK-13145 at 10/26/13 7:57 PM:
----------------------------------------------------------------------

@Gareth: Thanks for the reply.
I have issues configuring the phone on TCP only, it would not register.
sip.conf
{noformat}
[general]
tcpenable = yes
...
[100]
transport = tcp
{noformat}

This will produce the following log error:
{noformat}
[Oct 26 20:09:18] ERROR[32473]: chan_sip.c:17368 register_verify: 'UDP' is not a valid transport for '100'. we only use 'TCP'! ending call.
[Oct 26 20:09:18] NOTICE[32473]: chan_sip.c:30667 handle_request_register: Registration from '<sip:100 at 192.168.1.9>' failed for '192.168.1.10:49166' - Device not configured to use this transport type
{noformat}

The phone would not register. If I setup the transport to *tcp,udp*, it works but produces the re-transmission issues. Not sure if this is still valid on CCUM 9.3.4 software but form Cisco's perspective, port 5060 needs to be open on both TCP and UDP:
||*From (Sender)*|*To (Listener)*|*Destination port*|*Purpose*||
|Phone|Unified CM|5060 / TCP and UDP|Session Initiation Protocol (SIP) phone|

If you have any tips related to this issue, please let me know.
I updated the CentOS/Redhat 6 [Asterisk packages|http://rpm.axivo.com/] to 11.6.0 with your patch on *axivoplus* repository.

      was (Author: teck):
    @Gareth: Thanks for the reply.
I have issues configuring the phone on TCP only, it would not register.
sip.conf
{noformat}
[general]
tcpenable = yes
...
[100]
transport = tcp
{noformat}

This will produce the following log error:
{noformat}
[Oct 26 20:09:18] ERROR[32473]: chan_sip.c:17368 register_verify: 'UDP' is not a valid transport for '100'. we only use 'TCP'! ending call.
[Oct 26 20:09:18] NOTICE[32473]: chan_sip.c:30667 handle_request_register: Registration from '<sip:100 at 192.168.1.9>' failed for '192.168.1.10:49166' - Device not configured to use this transport type
{noformat}

The phone would not register. If I setup the transport to *tcp,udp*, it works but produces the re-transmission issues. Not sure if this is still valid on CCUM 9.3.4 but form Cisco's perspective, port 5060 needs to be open on both TCP and UDP:
||*From (Sender)*|*To (Listener)*|*Destination port*|*Purpose*||
|Phone|Unified CM|5060 / TCP and UDP|Session Initiation Protocol (SIP) phone| 

I updated the [Asterisk packages|http://rpm.axivo.com/] to 11.6.0 with your patch on *axivoplus* repository.

> [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-11.6.0.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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