It looks to me like calls from your Dial will route back to the sip-outgoing context and Dial again... it's loop. You'd really need to provide more logging information to advise further.
On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi <[email protected]> wrote: > > > 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 > > > _______________________________________________ > -- 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
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