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[tzury.by at reguluslab...]

Hi all,

I was following the instruction found at http://wiki.freeswitch.org/wiki/SIP_TLS
When I got to Step 4 I saw that instruction of editing the dial-string.
However, in my conf/dialplan/default.xml I did not found any matched entry .

Version I am using:
typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M)

thanks,
Tzury Bar Yochay

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[rupa at rupa.com]

hmm...  When I put that section in, I put the wrong filename.  It should be directory/default.xml.  I'll update the wiki.  Also, it is a valid configuration to support tls but not srtp.  I'll put a bit of a discussion in there talking about that. 

Setting sip_secure_media to true requires the endpoint do srtp.  There is no way (that I know of) to say "do srtp if possible but if not fallback to clear".  zrtp does fallback to clear if it can't negotiate keys.  But zrtp is supported by far fewer endpoints and no hardphones (as of yet).

On Wed, Jul 15, 2009 at 4:57 AM, Tzury Bar Yochay <tzury.by@reguluslabs.com> wrote:

[brian at freeswitch.org]

It tells you to edit conf/directory/default.xml not dialplan/
default.xml and put
<param name="dial-string" value="{sip_secure_media=${regex($
{sofia_contact(${dialed_user}@${dialed_domain})}|
transport=tls)},presence_id=${dialed_user}@${dialed_domain}}$
{sofia_contact(${dialed_user}@${dialed_domain})}" />

as the dial-string.

/b

On Jul 15, 2009, at 4:57 AM, Tzury Bar Yochay wrote:

[freeswitch-users at li...]

thanks allot, this was my mistake.
/Tzury