On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham <[email protected] > wrote:
> 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 > With many thanks for your reply , can you please confirm if the following scenario will work this way ? "My Asterisk on CentOS server is at @192.168.20.110 so I modified the sip.conf & extensions.conf as the followings to let the Asterisk to be both as client and server : Under sip.conf : --------------------- register => toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=sip-outgoing host=192.168.20.110 disallow=all allow=alaw Under extensions.conf : --------------------------------- [sip-outgoing] exten => _XXXXXXX,1,Dial(SIP/osaka/${EXTEN}) Then I issued the following : CLI>console dial 1234...@sip-outgoing But it didn't get through . Can you please do me favor and let me know what is my problem that I cannot get answer from this scenario at the Laboratory ? Thank you
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