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...eneral] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid="sip" <2222> username=sip host=188.208.12.37 accountcode=sip Thanks you all!!! Michelle ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/

Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/

...I'm not doing well the registration but I can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 is the IP of the SIP gateway. 2222 is one of the phones of the SIP Gateway...Anyone can help????Please! Thanks very very much Michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/

...OK From: To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37 CSeq: 102 BYE Via: SIP/2.0/UDP 188.208.12.237:5060 ;received=188.208.12.237 ;branch=z9hG4bK67231 48d Content-Length:0 7 headers, 0 lines Message is BYE ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/

...tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle <br><br> <b>Tu cuenta de correo gratuita <a href="http://mixmail.ya.com" target="_blank">Mixmail</a> con Antivirus y Antispam</b><br> Ya.com ADSL Home 24h, <a href="http://ir.ya.com/app/redir?o=2&prefix=isp_ya_com_&tgt=800&pck=10016&afiliado=autopromo" target="_blank"><b>Módem +...

...ntcode=sip</P> <P>Thanks you all!!!</P> <P>Michelle<BR></P> <br><br> <hr size="1"> <font size="2" face="Verdana, Arial, Helvetica, sans-serif"><strong>Tu cuenta de correo gratuita <a href="http://mixmail.ya.com" target="_blank">Mixmail</a></strong><br> Ya.com ADSL Home 24 h, <a href="http://acceso.ya.com/adslhome24h/" target="_blank"><b>Módem + Alta ¡Gratis!</b></a> </font>

...line? I think is in [general] section of the sip.conf and that I have to put: register => user:password@host:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/

I've tried to configure samba to lock files bewteen windows and linux but i couldn't i've read a lot of messages here, but trere is no one that have something about the file smb.conf. i have this in my global secction but i doesn't work [global] workgroup =3D GMC create mask =3D 0777 os level =3D 16 directory mask =3D 0777 hosts allow =3D

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

...;mso-ansi-language: EN-GB"><FONT face="Times New Roman"><FONT size=1>accountcode=sip<o:p></o:p></FONT></FONT></SPAN></P> <P>&nbsp;</P> <br><br> <b>Tu cuenta de correo gratuita <a href="http://mixmail.ya.com" target="_blank">Mixmail</a> con Antivirus y Antispam</b><br> Ya.com ADSL Home 24h, <a href="http://ir.ya.com/app/redir?o=2&prefix=isp_ya_com_&tgt=800&pck=10016&afiliado=autopromo" target="_blank"><b>Módem +...

Hi all. I am using the library Microsoft RTC to connect a windows software with * in a local network via SIP. In my sip.conf I have defined a 'friend' for this software with their username and their password. I don't want that nobody can be connected without the password. When I call to * it says: "Proxy Authentication Required" in the SIP debug. Please somebody can