[Freeswitch-dev] Note on using sipphone.com with FreeSWITCH

[Brian West]

Attention everyone:

	I have added some examples for setting up various sip providers into  
SVN under directory/default/ that you can use as templates to setup  
your connections to those providers.

One note that I must inform you all on is the fact that SIPPhone.com  
advertises _sip._tcp.proxy01.sipphone.com that maps to  
uap.sipphone.com port 443 which is 100% invalid and doesn't work.  So  
when you send an invite to sipphone it'll take 30 seconds for TCP to  
timeout and then fall back to UDP.

You have three options here:

1. Email SIPPhone and ask them to fix it.  Which I have done and yet  
to get a response about.
2. Disable TCP and only use UDP in oubound.xml profile that you  
register from for sipphone.
3. Do not use SIPPhone at all.

The bottom line is they shouldn't advertise an SRV record for TCP if  
they do not officially support it.

Thanks,
Brian West