AsteriskWin32 does have SIP server functionality, same as the linux version.
I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi <[email protected]>wrote: > > > On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < > [email protected]> 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 <[email protected]>wrote: >> >>> >>> >>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <[email protected]>wrote: >>> >>>> >>>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >>>> >>>> > >>>> > >>>> > >>>> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner <[email protected]> >>>> 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:[email protected]/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 > > > _______________________________________________ > -- 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
