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

Thu Dec 24 10:41:48 CST 2009

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 <motamedi24 at gmail.com> wrote:

>
>
> 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 --
>>>>>>
>>>>>> 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: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
>
>
> _______________________________________________
> -- 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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