[OpenSIPS-Users] Opensips + asterisk 1.4
Dani Popa
dani.popa at gmail.com
Wed Jul 17 13:39:28 CEST 2013
when you send a call in asterisk, do you see in asterisj cli that call hit
you callingcard context or it hit default context ?
On Wed, Jul 17, 2013 at 1:55 PM, Willian Mazzardo - SYSSVOIP <
willian at syssvoip.com.br> wrote:
> My a2billing context
>
> [callingcard]
>
> exten => _X.,1,DeadAGI(a2billing.php)
>
>
> Willian Mazzardo
> Depto TI - SYSSVOIP
> www.syssvoip.com.br
> 55 3537 2030
>
>
> 2013/7/17 Dani Popa <dani.popa at gmail.com>
>
>> what contex hit invite from opensips ?
>>
>>
>> On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP <
>> willian at syssvoip.com.br> wrote:
>>
>>> Hi Dani ... thanks ... i have for now insecure=very ... my asterisk
>>> version is 1.4... and this type of setting is for 1.6+
>>>
>>> Willian Mazzardo
>>> Depto TI - SYSSVOIP
>>> www.syssvoip.com.br
>>> 55 3537 2030
>>>
>>>
>>> 2013/7/17 Dani Popa <dani.popa at gmail.com>
>>>
>>>> set opensips peer to insecure=port,invite
>>>>
>>>>
>>>> On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP <
>>>> willian at syssvoip.com.br> wrote:
>>>>
>>>>> Hi Stephens... how do I do this?
>>>>>
>>>>> Willian Mazzardo
>>>>> Depto TI - SYSSVOIP
>>>>> www.syssvoip.com.br
>>>>> 55 3537 2030
>>>>>
>>>>>
>>>>> 2013/7/17 Stephen Vigus <svigus at gmail.com>
>>>>>
>>>>>> Hi Willian
>>>>>>
>>>>>> You most likely need to configure Asterisk to not authenticate SIP
>>>>>> requests coming from Opensips.
>>>>>>
>>>>>> Regards
>>>>>> Stephen
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP <
>>>>>> willian at syssvoip.com.br> wrote:
>>>>>>
>>>>>>> Hi all..
>>>>>>>
>>>>>>> I know this is a very simple scenario, all PSTN calls be routed to
>>>>>>> asterisk to do the billing job, but im having some problems, this is my
>>>>>>> scenario:
>>>>>>>
>>>>>>> Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247)
>>>>>>> ..... > PSTN
>>>>>>>
>>>>>>> Calls between sip clients on Opensips are working, but when I try to
>>>>>>> call over Asterisk, I have Proxy authentication problem.
>>>>>>>
>>>>>>> Here is my logs:
>>>>>>>
>>>>>>> Opensips: http://pastebin.com/SWpuRHku
>>>>>>> Asterisk: http://pastebin.com/6jp50LSS
>>>>>>>
>>>>>>> [opensips]
>>>>>>> host=10.1.1.2
>>>>>>> type=friend
>>>>>>> context=callingcard
>>>>>>> qualify=no
>>>>>>> insecure=very
>>>>>>> fromdomain=10.1.1.2
>>>>>>>
>>>>>>>
>>>>>>> Route: http://pastebin.com/mLgpXiNx
>>>>>>>
>>>>>>> Can someone help me on this?
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>> Willian Mazzardo
>>>>>>> Depto TI - SYSSVOIP
>>>>>>> www.syssvoip.com.br
>>>>>>> 55 3537 2030
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>
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>>>>>>
>>>>>
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>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Dani Popa
>>>>
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>>>>
>>>>
>>>
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>>
>>
>> --
>> Dani Popa
>>
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>>
>
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--
Dani Popa
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