> I am a newbie to * and I am having a problem which appears strange as I did > not find any mention of it anywhere in my search. > > Simply speaking, I have an external SIP proxy server which I am trying to > configure for incoming and outgoing calls from my asterisk installation. So > here is my configuration in sip.conf > > [general] > register => user:secret:[EMAIL PROTECTED]:8080 > > as long as I have just the above entry, I am able to receive incoming calls. > Now I would like to setup outgoing calls too. So I create a new section in > sip.conf > > [sipserverout] > type=peer > secret=secret > username=user > fromuser=user > fromdomain=sipserver.com > host=sipserver.com > port=8080 > context=default > > with the above configuration I can successfully dial out using > dial(SIP/[EMAIL PROTECTED]) > > but now when I call my incoming number, I get a busy or invalid number > signal. If I coment out sipserverout section, I could receive incoming calls > again. > > So I turned on sip debug on CLI. and it appears to me that the following is > happening. astreisk takes the incoming call and tries to match it with a > section with the same hostname. Now the reverse IP lookup on 109.147.41.48 > return sipserver.com (which is correct), so it is trying to send the call to > sipserverout which is essentially back to the same server where it came from > (Notice the statement "Found peer 'sipserverout'" in the sip debug logs > below). This creates an endless loop and the equipment at the other end > terminates the call. > > According to all the examples I have seen, my setup is the correct setup and > everyone seems to be using it. but it does not work for me. I am deperately > looking for a solution. Please help. > > I am using asterisk 1.2.0 beta 1 on FC1.
In very general terms, you probably want something like this in your sip.conf: [sipserver] type=friend secret=secret username=user fromuser=user fromdomain=sipserver.com host=sipserver.com port=8080 insecure=very canreinvite=no dtmfmode=inband context=from-sipserver disallow=all allow=ulaw For sip stuff, notice the use of type=friend and canreinvite=no. The use of the register statement (in this case) implies use of type=friend (for both incoming and outgoing calls). Then in extensions.conf, use something like this: exten => _1NXXXXXXXXX,3,Dial(SIP/sipserver/${EXTEN}) where SIP/sipserver is referring to the context [sipserver] in sip.conf. Did the folks at sipserver.com tell you to use port=8080? If not, remove that statement as the default for sip is port=5060. There are other ways to accomplish the same thing, so consider the above as only way to do it. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users