[Asterisk-Users] receiving calls from FWD

[John Fawcett]

Michiel van Baak wrote:

>On 15:14, Sat 13 Aug 05, John Fawcett wrote:
>  
>
>>I have successfully configured asterisk to make outgoing calls over FWD,
>>but cannot receive incoming calls. The console shows no messages,
>>even though an XTEN client on the same network has no problems receiving
>>incoming calls.
>>
>>This is the relevant part of sip.conf
>>
>>[general]
>>.....
>>register => 688426:xxxxxxxx at fwd.pulver.com/6000
>>
>>[fwd.pulver.com]
>>type=friend
>>username=688426
>>fromuser=688426
>>secret=xxxxxxx
>>dtmfmode=inband
>>host=fwd.pulver.com
>>port=5060
>>nat=yes
>>canreinvite=no
>>externip=194.185.53.47
>>localnet=192.168.1.0/255.255.255.0
>>
>>I orignally tried without the last two lines, then
>>added these and forwarded all ports from
>>the NAT to the asterisk server, but still no result.
>>
>>Is anyone with asterisk behind a NAT successfully receiving calls
>>from FWD? Can you give me any pointers on the above configuration?
>>
>>I realize that I could also try IAX which is supported by FWD, but
>>I'm having this problem with another SIP provider too, so I'd like
>>to get it working for SIP.
>>
>>    
>>
>
>Hi,
>
>Those 2 lines about externip and localnet should go before
>any register or phone/provider stanza.
>
>Check if you are registered to fwd/other sip provider.
>In asterisk CLI type: sip show registry.
>
>If you forwarded all ports that are stated in rtp.conf and
>the 5060 to the asterisk box it should work. At least it
>does here.
>  
>
thanks for the help. I moved the externip and localnet definitions
to the [general] section and now I can receive calls from FWD and
other sip providers.
Just for the list archives in case someone is looking to solve a
similar problem, here is some additional information which I
found useful: initially after applying the above solution I still
found that asterisk was not picking up the calls from FWD.
So I did a tcpdump -e -A to see the contents of the packets
being exchanged with the FWD sip server. I saw that asterisk
was sending an INVITE to the FWD sip server upon receiving
the initial incoming call request.
By adding insecure=invite to the [fwd.pulver.com] section
of sip.conf, this stopped asterisk issuing INVITE and asterisk
responded directly to the incoming calls.

John