[Ekiga-list] Can't call sip:500 at ekiga.net "proxy authentication required"

Damien Sandras dsandras at seconix.com
Fri Mar 9 14:47:37 UTC 2007


Le vendredi 09 mars 2007 à 14:39 +0000, Ken Booth a écrit :
> Damien Sandras wrote:
> >> I've setup an Asterisk server, with X-Lite 3.0 SIP softphones. I can 
> >> successfully call any PSTN number using my Asterisk dialplans, I can 
> >> call myself at sip:20 at myasteriskserver.co.uk and I can even call some 
> >> other companies e.g. sip:555 at didww.com
> >>
> >> My X-lite 3.0 config is set with a "Domain Proxy" of "target domain"
> >>
> >> However, whenever I try to make a test call e.g. to sip:500 at ekiga.net I 
> >> get the error "Proxy Authentication Required"
> >>
> >> I also get the same error when I try to call my sipgate account using 
> >> sip:1234567 at sipgate.co.uk, yet making a PSTN call to  (area 
> >> code)-123-4567 works (routing from PSTN to sipgate to my Asterisk server 
> >> to my X-lite SIP phone).
> >>     
> >
> > Ekiga.net only accepts calls from ekiga users, so you have to be
> > registered to ekiga.net so that it works. You can call other users
> > without being registered though, but not special services. I suppose it
> > is the same with sipgate.co.uk.
> >   
> Thanks, Damien.
> 
> I have registered an account with ekiga.net and tried to setup a 
> users.conf entry for my ekiga account on the Asterisk server.
> 
> My Asterisk server doesn't seem to get involved in the attempts to make 
> SIP calls, so how do I use my ekiga account my to make SIP calls, when 
> X-lite only allows me to register one account (and that has to be on my 
> Asterisk server)?
> 
> I have tried to use Asterisk as a proxy server in the X-lite config, but 
> then every call fails with "Caller not found"
> 

Ekiga can register several accounts. I do not know about XLite.
XLite setup questions should be directed to the XLite mailing list or
forums.

Just note that our Asterisk can register to all accounts for you, and
you can have your Ekiga softphone register only to Asterisk, which will
then route the call to the appropriate destination.

Asterisk setup questions should be directed to the Asterisk mailing
list.
-- 
 _      Damien Sandras
(o-      
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_  NOVACOM                 : http://www.novacom.be/
          FOSDEM                   : http://www.fosdem.org/
          SIP Phone             : sip:dsandras at ekiga.net
                       





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