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: [email protected] 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 >> -- ___________________________________________________ Play 100s of games for FREE! http://games.mail.com/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
