[asterisk-users] Softphone behind NAT issues

[Rob Schall]

No luck. Still no outbound sound.

Leonardo Kamache (Gmail) wrote:
> In [general] section:
>
> externip=your_extern_ip_address
> localnet=your_local_net/bits   i.e. 192.168.0.0/24
>
> Try this...
>
>
>
>
> On 6/12/07, Rob Schall <rschall at callone.net> wrote:
>>
>>  We are trying to use a softphone from a location that is behind a
>> firewall.
>> We are using x-lite as the softphone.
>>
>>  So far, we've been able to get the phone to register with the asterisk
>> server, and it can receive voice from the asterisk server (IE,
>> voicemail,
>> etc).
>>
>>  However, asterisk can't hear anything from the softphone. We have
>> used 2
>> different machines to test this on. We are watching the traffic and
>> noticed
>> that there doesn't appear to be any rtp traffic going back to
>> asterisk (this
>> is where we think the problem might be). The firewalls on both sides
>> have
>> ports 5060, 10000-20000 and 3478 (STUN) open.
>>
>>  Out conf files are:
>>  ------------------
>>  [sip.conf]
>>
>>  [general]
>>  context=incoming                ; Default context for incoming calls
>>  bindport=5060                   ; UDP Port to bind to (SIP standard
>> port is
>> 5060)
>>  bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0
>> binds to
>> all)
>>  srvlookup=yes                   ; Enable DNS SRV lookups on outbound
>> calls
>>  allow = all
>>
>>  [1000]
>>  nat=yes
>>  type=friend
>>  secret=Polycom
>>  context=internal
>>  host=dynamic
>>  canreinvite=no
>>  mailbox=1000 at default
>>  callerid=TESTUSER1 <1000>
>>
>>  -----------------
>>  [extensions.conf]
>>  exten => 1000,1,Macro(stdexten,1000 at default,SIP/1000)
>>  ----------------
>>
>>  [rtp.conf]
>>
>>  [general]
>>  rtpstart=12000
>>  rtpend=12005
>>  dtmftimeout=3000
>>
>>  What are we missing?
>>
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