[asterisk-users] Inquiry:Connect my Asterisk to external sip?

[David Cunningham]

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 <motamedi24 at gmail.com>wrote:

>
>
> On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <
> dcunningham at 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 <motamedi24 at gmail.com>wrote:
>>
>>>
>>>
>>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <fred at 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 <fred at 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:welcome at 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:welcome at 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:welcome at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091223/4aa23f7a/attachment-0001.htm