[Freeswitch-users] Configuration

[Brian West]

You revert the profiles too?

/b

On Jan 8, 2008, at 9:20 AM, Jonas Gauffin wrote:

> Back to basics. I've reverted to the default sip profile configs.
> I've now specified "192.168.1.102" as the domain (the FS ip). Both
> phones do now register using that ip.
> The registration goes OK.
>
> But when I call from one phone to another (both on the same lan as
> FS), I  only get audio one way.
> I call 1203 which brigdes to sofia/192.168.1.102/conny in the  
> dialplan.
> This must be the simplest test case, right?
>
> On Jan 8, 2008 2:00 PM, Brian West <brian.west at mac.com> wrote:
>> Btw if you put back the config to the default. It will work because
>> the ip is the domain and the packet don't loop thur your nat router
>> needlessly.
>>
>> /b
>>
>> Sent from my iPhone
>>
>> On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin at gmail.com>
>> wrote:
>>
>>
>>>>> * I only get audio one way when dialing from one registered UA to
>>>>> another, check attached log internal_call.log
>>>>
>>>> Chances are mydomain.com is your public IP right? and I suspect NAT
>>>> is
>>>> involved?  If so then you need to setup a profile for Natted phones
>>>> behind the same nat.  One profile for phones outside the nat to
>>>> register with FreeSWITCH ... and the public one listed below.
>>>>
>>>
>>> mydomain.com points on my public ip, correct. The phones and FS are
>>> behind a router (on the same lan).
>>> FS have loaded the three profiles (nat, default and outbound). I  
>>> have
>>> made no changes to them.
>>> Currently I have just tested with phones that is on the same lan as
>>> FS.
>>>
>>> here is a registration trans:
>>>
>>>  REGISTER sip:mydomain.com SIP/2.0
>>>  Via: SIP/2.0/UDP
>>> 192.168.1.123:
>>> 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93
>>>  From: <sip:tobbe at mydomain.com>;tag=66055002
>>>  To: <sip:tobbe at mydomain.com>
>>>  Call-ID: 58ee65fb0b696f5a370d839058387f2c at 192.168.1.123
>>>  Contact: <sip:tobbe at 192.168.1.123:5060>
>>>  CSeq: 53 REGISTER
>>>  Expires: 300
>>>  Authorization: Digest username="tobbe",realm="mydomain.com",[...
>>> some info stripped out
>>> ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5
>>>  User-Agent: CMI CM5K
>>>  Content-Length: 0
>>>
>>> tobbe has user_context set to "default" in conf/directory/default/
>>> tobbe.xml
>>>
>>>>
>>>>>
>>>>>
>>>>> * Inbound external calls do not work, i get [407][Proxy
>>>>> Authentication Required], check log external_call.log
>>>>
>>>> This is why you need to start another profile to accept calls from
>>>> the
>>>> outside world with authentication turned off.  That would fix that
>>>> problem.
>>>
>>> well. the nat profile exists and it has auth_calls set to false. but
>>> maybe it isnt used in my config?
>>>
>>>>
>>>> I honestly like having multiple UA's it solves many problems.
>>>>
>>>
>>> what do u mean by that? I'm using multiple UA's, all of them are
>>> currently on the same LAN as FS right now (testing enviroment), but
>>> later on, they may also be elsewhere.
>>> But currently I'm trying to get two UA's to be able to talk with  
>>> each
>>> other (as I said, the audio is just going one way).
>>>
>>> And I'm not saying that it's one UA to rule them all.. =) I'm just
>>> trying to understand this all new configuration how a profile is
>>> selected (for inbound/outbound external calls, internal calls,
>>> internal registration (on the same lan as FS) and external
>>> registration (not on the same lan as FS)).
>>>
>>> thanks,
>>> jonas
>>>
>>
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>
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