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

Wed Dec 23 23:16:30 CST 2009

On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham <dcunningham at voisonics.com
> 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 <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 --
>>>>>
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>>>>>   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 --
>>>>
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>>>>   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 --
>>>
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>>> 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:welcome at 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 1234567 at 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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