Gordon Henderson wrote:[color=blue]
> In article <f2e8ii$f3u$1$8300dec7@news.demon.co.uk>,
> Steve <user@mail.com> wrote:
>[color=green]
>> Hi Gordon,
>>
>> Thanks these info.. however, I stil have something wrong with it.
>> Here is the exact config I have :
>> I seemed to be connected to the server...
>> ButNoting is incoming
>>
>> If I add the line Exten => 0,1,goto(incoming,1009326,1)
>> and press the 0 => that's OK !!!
>> SO the script seems to be OK, except that it doesn't work if the phone
>> call is genereated by Sipgate.
>>
>> --------------------------
>>
>> Sip.conf
>>
>> [general]
>>
>> port = 5060
>> bindaddr = 0.0.0.0
>> allow=all
>> context=default
>> srvlookup=no ; could help if Phone using DNS
>> externip=xx.xx.xx.xx
>> localnet=192.168.3.0/255.255.255.0
>>
>> register => 1009326:xxxx@sipgate.co.uk/1009326
>>
>> [sipgate]
>> type=friend
>> secret=xxxxx
>> username=1009326
>> fromuser=1009326
>> fromdomain=sipgate.co.uk
>> host=sipgate.co.uk
>> insecure=port,invite
>> qualify=yes
>> nat=no
>> context=incoming
>> disallow=all
>> allow=alaw
>> allow=g726
>> allow=gsm[/color]
>
> As you have externip= and localnet= settings above, you probably
> want nat=yes here. The box I cut & pasted that off isn't behind a NAT
> firewall/router.
>
> Remember to connect to asterisk and set verbose 9999 which will help
> you debug.
>
> asterisk -r
> set verbose 9999
>
> then call your sipgate number from another phone.
>
> And you can use sip show peers to make sure you're actually peering
> with Sipgate - if nothing else it'll make sure your register statement
> is working OK.
>[color=green]
>> EXTENSION.CONF
>>
>> [general]
>> ;
>> static=yes
>> writeprotect=no
>> Static=yes ; DialPlan can't be modified by Asterisk
>> clearglobalvars=no
>> priorityjumping=no
>> autofallthrough=yes
>> [globals]
>>
>> GROUP=SIP/120&SIP/121
>> INCOMING1=1009326
>>
>> [incoming]
>>
>> exten => ${INCOMING1},1,Noop(Incoming SIP call from ${CALLERID(all)} @
>> sipgate calling ${EXTEN})
>> exten => ${INCOMING1},2,SetCallerID(From SipGate <${CALLERID(num)}>)
>> Exten => ${INCOMING1},2,Set(TRIES=0)
>> Exten => ${INCOMING1},3,ringing(3)
>> Exten => ${INCOMING1},4,background(priv-introsaved)
>> Exten => ${INCOMING1},5,background(queue-thankyou)
>> Exten => ${INCOMING1},n,Dial(SIP/121,10,r)
>> Exten => ${INCOMING1},n,playback(vm-nobodyavail)
>> Exten => ${INCOMING1},n,Dial(${GROUP},10,mr)
>> Exten => ${INCOMING1},n,playback(tt-allbusy)
>> Exten => ${INCOMING1},n,wait(2)
>> Exten => ${INCOMING1},n,goto(${INCOMING1},7)
>> ;Exten => ${INCOMING1},n,voicemail(u300@default)
>> Exten => ${INCOMING1},n,congestion[/color]
>
> Er - interesting call-flow there :)
>
> You should Answer() the call at some point too, preferably before you
> try to send audio back up the line (which ringing(3) will do) although
> it should auto-answer the channel, I've found that it's always best to
> do it explicitly.
>
> If you start to use goto's, then you
> really ought to start to use labels and the 'n' priority. Eg.
>
> exten => 123,1,Answer()
> exten => 123,n(loop),Playback(haha)
> exten => 123,n,Wait(2)
> exten => 123,n,Goto(loop)
>
> It saves counting lines and re-numbering when you add/remove instructions.
>
> Gordon[/color]

Hi Gordon,

That's works now..
The probelm was modem router...
I placed the server on DMZ area, and everything is OK..

This is my main problem. I try yto be able to be connected from outsite
the house, and I can't connect...

Can you connect to your server from outside ?
If so, what Modem router do you have ??

THanks and regards,

Steve