[Asterisk-Users] RE:IConnecthere dial out problems

[John Voss]

I finally got my issue resolved. It actually had nothing to do with my SIP.conf file. The problem was how I was trying to set the callerid in my extensions.conf file.

Anyway, do you have other voip providers that are working?
Do incoming calls work at all prior to timing out?
Are you NAT'ed, or are you behind a broadband router?

If you are NAT'ed and you haven't already configured * for it you may have issues like this.

(from http://www.voip-info.org/wiki/view/tips)
When sip is behind a NAT do not forget to specify: 

in sip.conf 

[general] 
nat=yes 
externip = X.X.X.X 
fromdomain = yourdomain.com 
localnet = 192.168.X.0/255.255.255.0

I choose to use externhost = yourdomain.com instead of externip since most broadband providers use DHCP and you address can change. I registered a domain with no-ip.com (it's free) and use that in place of yourdomain.com. They also have a client that you can load on a windows box that keeps track of your external ip and updates your domain if your ip ever changes. That way I don't have to worry about it.

You may need to add srvlookup to your sip.conf to allow name resolution if you use externhost instead of externip:
srvlookup=yes


You will also need to setup port forwarding on your broadband router/firewall:

(from http://www.voip-info.org/wiki/view/NAT+and+VOIP)
SIP signaling: Ports 5060 to 5070 
RTP audio: Ports 8766 to 35000 

I only forward the following listening ports (read comments in the wiki for this)
SIP signaling: Port 5060
RTP audio: Ports 1000 - 2000 (you can restrict this in RTP.conf)


Try these wiki pages for more info:
          http://www.voip-info.org/wiki-Asterisk+config+rtp.conf
          http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
          http://www.voip-info.org/wiki/view/NAT+and+VOIP

----- Original Message ----- 
From: "Dennis Gilmore" 
To: asterisk-users at lists.digium.com 
Subject: Re: [Asterisk-Users] RE:IConnecthere dial out problems 
Date: Wed, 7 Dec 2005 21:49:01 -0600 


Once upon a time Wednesday 07 December 2005 8:42 pm, John Voss wrote: 
> Your SIP.conf file looks much different than mine. I'll give it a try. 

Hope mine helped 

> [iconnect] 
> type=friend 
> secret= 
> username= 
> host=213.137.73.140 ;sipauth.deltathree.com 
> permit=213.137.73.140/255.255.255.0 
> permit=208.170.168.0/255.255.255.0 
> disallow=all 
> context=incoming 
> allow=gsm 
> allow=ulaw 
> allow=alaw 
> allow=G726 
> insecure=very 
> nat=Yes 
> canreinvite=no 
> 
> I don't know what your register line looks like in your SIP.conf. This is 
> mine. 
> 
> register => ::@213.137.73.140:5060 
> 
> I was unable to receive calls until I added the insecure=very line. 
mine is register => ::@natrelay.deltathree.com 

i can receive incomming calls for a little while after a reload but after 
some timeouts incomming calls stop 
-- 
Dennis Gilmore, RHCE 
http://www.ausil.us 
<< 2.dat >> 

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