On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < dcunning...@voisonics.com> wrote:
> Hadi, > > You could use Asterisk as a sip server, it's installable on Windows. > > Using "sip set debug on" might help you with the "Host '192.168.0.139' does > not implement 'REGISTER'" problem. > > > On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi <motamed...@gmail.com>wrote: > >> >> >> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <f...@teamforrest.com>wrote: >> >>> >>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >>> >>> > >>> > >>> > >>> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner <f...@teamforrest.com> >>> wrote: >>> > >>> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: >>> > >>> > > Dear All >>> > > I have an application that calls for my Asterisk sip to be connected >>> to an external sip server for voip routing . Please be informed that my >>> Asterisk sip is at @192.168.0.2 and the external sip is at @ >>> 192.168.0.139 . To this end , I modified my sip.conf & extensions.conf >>> as the followings : >>> > > Under sip.conf : >>> > > --------------------- >>> > > [general] >>> > > register => toronto:welc...@192.168.0.139/osaka >>> > > [osaka] >>> > > type=friend >>> > > secret=welcome >>> > > context=osaka_incoming >>> > > host=dynamic >>> > > disallow=all >>> > > allow=alaw >>> > > [6672019] >>> > > type=friend >>> > > host=dynamic >>> > > context=phones >>> > > >>> > >>> > Try this: >>> > >>> > [general] >>> > register => toronto:welc...@osaka >>> > >>> > [osaka] >>> > type=friend >>> > username=toronto >>> > authname=toronto >>> > secret=welcome >>> > context=osaka_incoming >>> > host=192.168.0.139 >>> > disallow=all >>> > allow=alaw >>> > >>> > Although your error shows the other server does not allow register. >>> What is the other server? >>> > >>> > ---fred >>> > http://qxork.com >>> > >>> > >>> > Thank you for your reply . The other server is not an Asterisk sip >>> server . It is a sip server inside a softswitch from a third party vendor . >>> As the external sip server man is asking me to disable for the >>> authentication at the first stage , can you please let me know how can I >>> disable for the authentication at this stage (when the calls get through I >>> will enable it again) ? >>> > Thank you in advance >>> > >>> >>> [general] >>> ;register => toronto:welc...@osaka >>> >>> [osaka] >>> type=friend >>> ;username=toronto >>> ;authname=toronto >>> ;secret=welcome >>> context=osaka_incoming >>> host=192.168.0.139 >>> disallow=all >>> allow=alaw >>> >>> >>> ---fred >>> http://qxork.com >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> Thank you for your reply . Please be informed that I want to simulate this >> case in the Laboratory , i.e. connecting my Asterisk sip to external sip >> server with the guidelines you sent me . Can you please propose for an Voip >> application sw that I can install on my MS Windows client and plays the >> external sip server side role ? It seems that Skype is not suitable for this >> case as it cannot be configured to play the role of external sip server . >> Thank you in advance >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > David Cunningham > Voisonics > IVR development, VOIP consultancy > http://voisonics.com/ > US toll-free: +1 888 842 2720 > UK: +44 (0) 20 3411 5024 > Australia: +61 (0) 2 9037 2180 > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > I downloaded & installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you
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