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

David Cunningham dcunningham at voisonics.com
Wed Dec 23 04:22:51 CST 2009


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
>>
>>
>>
>>
>>
>>
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>>
>
>
>
> 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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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>



-- 
David Cunningham
Voisonics
IVR development, VOIP consultancy
http://voisonics.com/
US toll-free: +1 888 842 2720
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