[Asterisk-bsd] SIP Clients - No voice.

Olle E. Johansson oej at edvina.net
Tue Aug 31 15:35:21 CDT 2004


Jefferson Carvalho wrote:

> Hi Josansson,
> 
> I'm using FreeBSD 5.2.1-RELEASE / Asterisk-1.0.RC2
> Those clients aren't behind nat.
> They're on the same net!
Remove nat=yes and re-instate the disallow= line.
Try again.

/O

PS. And read the docs for the SIP channel on the wiki - there's a lot of docs
over there. http://www.voip-info.org :-)

> 
> 
> Regards,
> 
> -Jefferson
> 
> 
> Olle E. Johansson escreveu:
> 
>> Jefferson Carvalho wrote:
>>
>>> Richard,
>>>
>>> I'm using at this first momment only X-Lite as Sip client.
>>> I'm waiting for Sipura and Grandstream devices
>>> arrive from USA.
>>> I'll have to put a TDM04B ( 2FXO/2FXS  ) to
>>> work .. too. (  My PABX is a siemens HiPath )
>>> So ... i'll have so much work..!
>>
>>
>> Start with connecting everything without having NAT between clients 
>> and Asterisk.
>> When you have that working, start reconfiguring for the NAT...
>> Start simple. NAT traversal is not simple.
>>
>>>
>>>  >>>>>>A bit of my sip.conf.
>>>
>>> [1260]
>>> type=friend
>>> username=1260
>>> mailbox=2500
>>> context =default
>>> secret=jeff
>>> callerid="Jefferson Carvalho" <1260>
>>> host=dynamic
>>> nat=yes                       ; X-Lite is behind a NAT router
>>> canreinvite=no                ; Typically set to NO if behind NAT
>>
>>
>>
>>> ;disallow=all
>>
>>
>> You need to enable this before you have any allow= lines! Remove the 
>> semicolon.
>>
>>> allow=gsm                     ; GSM consumes far less bandwidth than 
>>> ulaw
>>> allow=ulaw
>>> allow=alaw
>>>
>>
>> What kind of BSD system are you using?
>>
>> /O
>>
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>>
> 
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-- 
Olle E. Johansson, Edvina.net AB, oej at edvina.net
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