Thanks Alejandro, you were right it was just a NAT problem ! i add a stun server in the phone configuration and it works :)
2010/11/19, Alejandro Imass <[email protected]>: > On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier <[email protected]> wrote: >> Hello, >> >> I have a Sip phone (Siemens C470IP) which works perfectly with >> different VoIP providers (iptel, betamax, ovh...). It also worked well >> with my testing server (ubuntu and inside the LAN). >> > > I am assuming you mean Asterisk on Ubuntu inside the LAN > >> But now the problem i have is that the hardphone doesn't connect to my >> dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing >> is that ekiga can connect to the same asterisk server with the same >> SIP account. >> > > Is this outside the LAN? > Is there NAT in between? > SIP is a pain in the ass with NAT, so it's the only thing I can think > of. Usually in my experience it's the other way around! Ekiga is the > one that doesn't work and tends to be very quirky (takes a long time > to quit, has strange registration quirks, etc.), I mean when compared > to HW SIP device. > >> Here is a part of my sip.conf : >> >> [general] >> dtmfmode=auto >> language=fr ; pour les messages lus par asterisk >> disallow=all >> allow=ulaw >> allow=alaw >> allow=speex >> >> [siemens] >> type=friend >> context=interne >> host=dynamic >> secret=xxxxxxxx >> >> When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some >> information. It seems that asterisk receives the rengistration request >> but doesn't answer to it. Here are the logs : >> http://server.chab.info/Registration_logs_ip_phone.txt >> >> Using Ekiga with the same SIP account (name is siemens) and from the >> same physical location works well : >> http://server.chab.info/Registration_logs_ekiga.txt >> >> I didn't change anything about asterisk config (except sip.conf and >> extensions.conf). >> If you have any idea, please share it with me, i really don't to do to >> fix this problem... >> Thanks in advance ! > > The only thing I can think of are NAT issues with SIP. If you are in > fact NATing try the Siemens phone to a direct IP to the server (no > NAT, firewall, etc.) and see. > > -- > Alejandro Imass > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
