[Freeswitch-users] Yealink T48G and TLS
Iskren Hadzhinedev
iskren.hadzhinedev at ikiji.com
Sat Apr 26 02:57:42 MSD 2014
On Friday 25 April 2014 21:19:01 Yehavi Bourvine wrote:
> Hi,
Hello, thank you for the advice!
>
> There might be two issues here:
>
> 1. Yealink had a bug in TLS omplementation and FreeSwitch. This has been
> fixed about a month ago, so make sure you are using the latest firmware.
There was an update from 18 April that I just applied but it had no effect on the issue. I did a
factory reset and reconfigured the phone after the update but nothing changed.
> 2. Make sure you set:
> <action application="set" data="sip_secure_media=true"/>
> <action application="export" data="nolocal:sip_secure_media=true"/>
> <action application="set" data="rtp_secure_media=true"/>
> <action application="export" data="nolocal:rtp_secure_media=true"/>
>
I didn't have the ones with "nolocal". I added them however, but it still didn't make a difference.
What I noticed is that the SRTP is working fine, even when I'm using TCP/UDP for SIP, it sends the
correct messages and the audio is working just fine (all other phones are set to mandatory SRTP
so they can't establish a call when SRTP is not working on the other end). However I tried TLS
with unencrypted media and that one failed too. I thought it's a NAT issue so I enabled a stun
server on the phone, but strangely enough, that had no effect either. I'll try updating to latest
git as this installation is a bit outdated, but since everything else is working I'm pretty confident
it's not FreeSWITCH's fault.
Thanks for the tips!
> before calling the phone (some of the above lines might be superfluous, but
> I didn't bother to check it).
>
> Regards, __Yehavi:
>
>
> 2014-04-25 16:38 GMT+03:00 Iskren Hadzhinedev <iskren.hadzhinedev at ikiji.com>
> > Hello everyone,
> >
> > Just got a couple of new Yealink T48G phones and I am having a couple of
> > rather weird (at least for me) issues with them.
> >
> >
> >
> > Whenever I enable TLS authentication on the phones, they register with the
> > FreeSWITCH box but there's no media on the
> >
> > outbound calls and I'm unable to get any incoming calls at all. If I
> > switch the authentication protocol to TCP or UDP it's all
> >
> > working good. I tried enabling/disabling SRTP for all 3 protocols and it
> > kept the behaviour consistent.
> >
> >
> >
> > All calls made are local (registered to the same FreeSWITCH box) and only
> > these phones have any issues with calls.
> >
> > I can call someone from the Yealink and then add another person in a 3-way
> > conference.
> >
> > That way the two remote parties can hear eachother, but the Yealink is
> > dead silent.
> >
> >
> >
> > Here's the profile on which all phones are connected:
> >
> >
> >
> > <profile name="local">
> >
> > <domains>
> >
> > <domain name="all" alias="true" parse="false"/>
> >
> > </domains>
> >
> > <settings>
> >
> > <param name="debug" value="0"/>
> >
> > <param name="sip-trace" value="no"/>
> >
> > <param name="sip-capture" value="no"/>
> >
> > <param name="watchdog-enabled" value="no"/>
> >
> > <param name="watchdog-step-timeout" value="30000"/>
> >
> > <param name="watchdog-event-timeout" value="30000"/>
> >
> > <param name="log-auth-failures" value="true"/>
> >
> > <param name="forward-unsolicited-mwi-notify" value="false"/>
> >
> > <param name="rfc2833-pt" value="101"/>
> >
> > <param name="sip-port" value="5060"/>
> >
> > <param name="dialplan" value="XML"/>
> >
> > <param name="liberal-dtmf" value="true"/>
> >
> > <param name="dtmf-duration" value="2000"/>
> >
> > <param name="inbound-codec-prefs" value="SILK,OPUS,G722,PCMU,PCMA,GSM"/>
> >
> > <param name="outbound-codec-prefs" value="PCMU,PCMA,GSM"/>
> >
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