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
