Gabriel,

I had the similar issues with Polycom phones. Stay away from the web configuration. 

All you need to setup the FTP or TFTP server and use the configuration files. I am currently using polycom phones in a Multi-Tenant [SRV Records] environment without any issue afterstart use the configuration files.



Thanks
Lloyd


On Sat, Jan 29, 2011 at 2:25 PM, Gabriel Gunderson <gabe@gundy.org> wrote:
All,

I have a multi-tenant FreeSWITCH server up and running.  We have all
the proper SVR records setup for each domain and everything works
great.  Most SIP clients Just Work (tm), however, the new Polycom
phones I have don't seem to auth properly with domains.  I've tried
every combination of settings (via the web config, I haven't setup a
TFTP server yet).  The issue is that auth has the domain twice, like
so: user@domain.tld@domain.tld.

It seems like the Internet has threads of discussion where others
express frustration with this, but I can't seem to find a documented
solution.  It's maddening because every other User Agent Client works
swimmingly and no matter what I've tried, I can't get the right
combination of settings on the Polycom.  I love the phones, but this
is getting old :)

Anyway, any pointers for me?

Thanks,
Gabe

sofia.conf:
*************************************************************
[SNIP]

<domains>
 <domain name="all" alias="true" parse="false" />
</domains>
<settings>
 <param name="log-auth-failures" value="true" />
 <param name="forward-unsolicited-mwi-notify" value="false" />
 <param name="context" value="public" />
 <param name="sip-port" value="$${internal_sip_port}" />
 <param name="dialplan" value="XML" />
 <param name="inbound-codec-prefs" value="$${global_codec_prefs}" />
 <param name="outbound-codec-prefs" value="$${global_codec_prefs}" />
 <param name="rtp-ip" value="$${internal_ip}" />
 <param name="sip-ip" value="$${internal_ip}" />
 <param name="hold-music" value="$${hold_music}" />
 <param name="record-path" value="$${recordings_dir}" />
 <param name="record-template"
value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
/>
 <param name="auth-calls" value="true" />
 <param name="ext-rtp-ip" value="auto-nat" />
 <param name="ext-sip-ip" value="auto-nat" />
</settings>

[/SNIP]

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