[Freeswitch-users] 480 Temporarily Unavailable[MANDATORY_IE_MISSING]

Jurijs Ivolga jurijs.ivolga at gmail.com
Thu Mar 31 18:30:14 MSD 2016


Hi Rutu,

Just a shot in a dark. :)

Did you added sip peer record in sip.conf, something similar to:

[freeswitch]
type=peer
host=IP_ADDRESS_OF_FREESWITCH_SERVER
username=HOSTNAME.DOMAIN.COM
port=5080
fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
secret=BOOTH_WAY_PASSWORD

With kind regards,


Jurijs

On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hi Michael
>
> Thank you for your assistance.
>
> We are trying to make call between two freeswitch users.
>
> One user is registered in softphone and other is registered in Asterisk by
> register string but asterisk user cannot authenticate properly and
> freeswitch directly send 480 Temporary Unavailable.
>
> In sip profile we set challenge-realm= auto_to also tried with auto_from
> but still not getting success.
>
> I am attaching some more logs if anyone can guide in right direction.
>
> --
> Thanks,
> Rutu Patel
>
>
>
> On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc at freeswitch.org>
> wrote:
>
>> Try a gateway. Look in conf/sip_profiles/external/example.xml for a
>> heavily-commented sample. For more detailed discussion see this nice doc:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
>>
>> -MSC
>>
>> On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hi,
>>>
>>> Thank you for reply.
>>>
>>> We have registered extension of Freeswitch in Asterisk as below
>>>
>>> register=>test-ext:secret at 192.168.1.64/test-ext
>>>
>>> ​We want to send call from Freeswitch to Asterisk​.
>>>
>>> What authentication parameters we need to set in Freeswitch ?
>>>
>>> --
>>> Thanks,
>>> Rutu Patel
>>> On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> You need to supply auth credentials to Asterisk because its set to
>>>> authenticate calls.
>>>> The best thing to do would be to make a gateway pointing to Asterisk
>>>> with all the credentials and call it using that.
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <
>>>> jurijs.ivolga at gmail.com> wrote:
>>>>
>>>>> Hi Rutu,
>>>>>
>>>>> In logs I can see that user-agent Asterisk is sending INVITE to
>>>>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This
>>>>> a bit differs from what you explained in your mail...
>>>>>
>>>>> First think what pop up in my mind is that you need to configure
>>>>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my
>>>>> first guess...
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>>>>
>>>>> With kind regards,
>>>>>
>>>>> Jurijs
>>>>>
>>>>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel at inextrix.com>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> We are sending call to asterisk PBX but we are getting 480
>>>>>> Temporarily Unavailable immediately. from freeswitch server.
>>>>>>
>>>>>> Call flow is as below:
>>>>>>
>>>>>> User => opensips => FS => PBX,
>>>>>>
>>>>>> but call not sending out from FS and drop by FS.
>>>>>>
>>>>>> Here I am attaching both sip logs and freeswitch logs.
>>>>>>
>>>>>> Can anyone please help to sort out the issue.
>>>>>> --
>>>>>> Thanks,
>>>>>> Rutu Patel
>>>>>> <http://www.inextrix.com>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
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>>>>>> http://www.cluecon.com
>>>>>>
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>>>>>> UNSUBSCRIBE:
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>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
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>>>>> http://www.cluecon.com
>>>>>
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>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>>
>>>>http://freeswitch.org/http://cluecon.com/>>>> http://twitter.com/FreeSWITCH
>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
>>>> <http://freeswitch.org/g+>*
>>>>
>>>> ClueCon Weekly Development Call
>>>> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
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>>>>
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>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
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>>
>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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