[Freeswitch-users] SIP TLS (and SRTP)

[Brian West]

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:

> 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