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
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