sipgate

[sjeffrey]

Hi,

I am using asterisk 1.07, and have set up an account with sipgate for my out going calls. If i follow what it on the on the sipgate site (below) then neither incoming or outgoing works, but if i change it so the register doesnt have the extension on it then i can get incoming to work (but still not outgoing).

Its strange because also with my method it works incoming without changing any of my firewall forwarding... is it using UPNP?


What it says on the sip site and doesnt work,

In the sip.conf:

register => 9888...ass@sipgate.co.uk/9888


[sipgate]
type=friend
username= 9888...
secret=pass
host=sipgate.co.uk
fromuser=9888...
fromdomain=sipgate.co.uk
nat=yes
authuser=9888...
dtmfmode=info
context=inbound
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw

In extensions.conf


exten => _7.,1,SetCallerID(01..9888...)
exten => _7.,2,Dial(SIP/${EXTEN:1}@sipgate,20,tr)
exten => _7.,3,Congestion
exten => _7.,4,Busy
exten => _7.,5,Hangup

[inbound]
exten => 9888...,1,Answer
exten => 9888...,2,Dial(sip/stuart)




And if i change sip.conf register line to just

register => 9888...ass@sipgate.co.uk


and extensions.conf inbound to

[inbound]
exten => s,1,Answer
exten => s2,Dial(sip/stuart)



Then i get inbound only


Does it matter where you place the reister line... ie i have mine under the general bit, should it be at the top?

Thanks

stuart

[sjeffrey]

Well i've managed to figure out the what is causing the problem. I took the offending asterisk box over to my friends and plugged into her broad band and bingo all workeed (again no need for any port forwarding).

So i guess the problem is that my router isnt forwarding the sip command port through?? ie when a call comes in for a specific extension its not getting the bit that tells it its an extension.

Its a solwise 715 router... but i doubt any one knows it,

Any way i have tried forwarding port 5060 & 5061 and that doesnt help.

Any suggestions?

[mberlant]

When you were over at your friend's home were you able to receive calls via the Asterisk? I would guess that you were not.

Asterisk does not have a STUN client, so you must manually either place the Asterisk in your DMZ or manually forward all of the necessary port ranges through the router. This includes 5060 for inbound SIP, 10000-20000 (or some narrower range that you have defined in your Asterisk) for inbound RTP, 22 for inbound SSH if you use that for management, and some others for additional features or service that you may be implementing.

In addition to this, if your Solwise 715 router is a Symmetric NAT router (about 2% of routers are), then you will need to do all of the above even to support outbound calling.

[sjeffrey]

Yeah when i was at my friends both incoming and outgoing calls were possible.. I didnt put the asterisk in the dmx and i didnt need to open any ports! It just worked.... I was suprised but it did. She has a Dlink Nat gateway. Cant remember the model number.

It would also work using Xlite without making any changes tot the firewall, or using a stun server.

How do i find out the port range that needs forwarding... is there a RTP.conf?

[mberlant]

I think it's in the sip.conf near the top. I'm not near a console to confirm this, so this is just off the top of my head.