[OpenSIPS-Users] Opensips + asterisk 1.4

[Willian Mazzardo - SYSSVOIP]

No ... just sip messages, and stops at Proxy Authentication Require.

Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030


2013/7/17 Dani Popa <dani.popa at gmail.com>

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