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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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:[EMAIL PROTECTED] >> 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 > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
