[Freeswitch-users] mod_rtmp login

[Jack]

Ricardo,
  It could be that your hard phone is coming in on port 5080 instead of 
port 5060 and ending up in your public dialplan instead of your default 
dialplan.
Jack

On 11/1/2011 12:43 PM, rsaavedra at ecogizmos.com wrote:
> Thank you,
> I'm using on the flex client the user 1002 at 1.2.3.4, the user 1000 at 1.2.3.4
> on a sip hard phone (aastra 480).
>
> My problem in this moment is: the user 1002 can hear but the user 1000
> can't hear.
>
> Best regards,
>
> Ricardo Saavedra
>
>
>
>> That would be correct if your FreeSwitch server IP address is 1.2.3.4
>>
>>
>> On 10/26/2011 10:04 AM, rsaavedra at ecogizmos.com wrote:
>>> Thank you,
>>>
>>> so in the flex client I will write the user like 1000 at 1.2.3.4
>>>
>>> Best regards,
>>>
>>> Ricardo
>>>
>>>> Ricardo,
>>>> If your 1000 is default it is probably set up with
>>>> <variable name="*user_context*"value="*default*"/>
>>>>
>>>>
>>>> if you set the param in rtmp.conf.xml to match the context  and then do
>>>> a reloadxml or restart freeswitch it may authorize for you.
>>>> <param name="context" value="default" />
>>>>
>>>> Hope that helps!
>>>> Jack
>>>>
>>>>
>>>>
>>>> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote:
>>>>> Hello,
>>>>>
>>>>> I'm trying to use mod_rtmp but everytime I tried on the console I see
>>>>> User
>>>>> not authorized.
>>>>> I was trying with user:1000 and password:1234.
>>>>>
>>>>> In the file rtmp.conf.xml I have all by default:
>>>>>     <param name="bind-address" value="0.0.0.0:1935" />
>>>>> <param name="context" value="public" />
>>>>> <param name="dialplan" value="XML" />
>>>>> <param name="auth-calls" value="true" />
>>>>> <param name="buffer-len" value="50" />
>>>>> <param name="chunksize" value="512" />
>>>>>
>>>>> Best regards,
>>>>>
>>>>> Ricardo Saavedra
>>>>>
>>>>>
>>>>>
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>>>>>
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>
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