[Asterisk-Users] NAT problems

Bartosz Wegrzyn junk at lexon.ws
Thu Aug 5 15:09:50 MST 2004


Thanks for the helpful information.
I must say that although I was using the port forwarding I had the
nat=yes set on.  I did that due to the fact that my asterisk didnot work
without this setting turned on. I dont know why.  Also, I was told by
other people that this must be on even with port forwarding. Today, I
changed it to off mode and now it works.  I still dont know how it will
relate to my droped calls.  I will have to test it.  Usually I need to
call into my box around 10-15 times, then wait some time to cause the
issue.  I will try testing it and let you know.  So far it works even the
IP after SIP/ is funny.

Bart,


> ----- Original Message -----
> From: "Bartosz Wegrzyn" <junk at lexon.ws>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, August 05, 2004 2:00 PM
> Subject: Re: [Asterisk-Users] NAT problems
>
>
>> I didnot use that magic name, because lately there was a big discution
>> regarding my post.  I will try to be as precise as as I can now.
>>
>> This is my sip conf file
>>
>> [general]
>> externip=my DDNS Domain name
>> bindaddr = 0.0.0.0
>> port=5060
>> localnet=192.168.1.0/255.255.255.0
>> disallow=all
>> allow=ulaw
>> context=from-broad
>> dtmfmode=inband
>> register => 7734660101:mysecret at sip.broadvoice.com
>> tos=0x18
>> srvlookup=yes
>> nat=yes
>> insecure=yes
>>
>> [Broadvoice]
>> type=peer
>> username=7734660101
>> fromuser=7734660101
>> secret=mysecret
>> host=sip.broadvoice.com
>> context=sip
>> fromdomain=sip.broadvoice.com
>> canreinvite=no
>> dtmfmode=inband
>> nat=yes
>>
>> [broadvoice-incoming]
>> type=peer
>> dtmfmode=inband
>> host=147.135.8.128
>> context=from-broad
>> qualify=yes
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>> nat=yes
>>
>> [broadvoice-incoming2]
>> type=peer
>> dtmfmode=inband
>> host=147.135.0.128
>> context=from-broad
>> qualify=yes
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>> nat=yes
>>
>> What is happening is that when this IP change occurs,
>> asterisk answers incoming call, but on the other side the caller does
>> not
>> here anything (only 2-3 rings) and then it goes directly to the provider
>> voicemail. When I look at the asterisk console while it is happening the
>> asterisk executes everything in the context till the end.
>>
>> Bart,
>
> Hmmmm if BV is answering voicemail that means that * is loosing
> registration
> or cannot contact the SIP proxy, not necessarily NAT related...
>
> One thing you should do, if you are not doing port forwarding for 5060 and
> your RTP ports, be sure you're doing that, it fixes a lot of things...
> right
> now since * does not do STUN it dosen't (at least not to me) seem to
> handle
> NAT very well...
>
> if you are using port forwarding, make sure you set NAT=NO or NAT=NEVER
> under all of the broadvoice contexts and in [general].
>
> the insecure=yes belongs in the [broadvoice-incoming] contexts, but you
> don't really need that since you've created 2 separate contexts...
>
> In the [broadvoice] context, you're gonna want to change it from saying
> host=sip.broadvoice.com to host=147.135.8.129. The reason for this is that
> sip.broadvoice.com resolves to both 147.135.8.129 and 147.135.0.129 and
> 0.129 is not accepting connections right now... maybe never, I don't know
> BV's plans... *I believe that this is the reason why your calls are not
> working some of the time...*
>
>
>     -Chris
>
>
>
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