[OpenSIPS-Users] Opensips + asterisk 1.4

Willian Mazzardo - SYSSVOIP willian at syssvoip.com.br
Wed Jul 17 12:55:29 CEST 2013


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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>> --
>>> Dani Popa
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Dani Popa
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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