Bruce, I had a similar problem when I was setting [EMAIL PROTECTED] sometime in 2005 and the following info helped me to figure out this problem: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip Hope this is useful Elliott
-----Original Message----- From: Bruce Nik [mailto:[EMAIL PROTECTED] Sent: Thursday, March 22, 2007 8:42 PM To: [email protected] Subject: [on-asterisk] SIP device from outside can't connect ; connection is provided by Allstream Hey guys, Well, from the previous posts I see that no one had any problems with Allstream connection so let me post what I have here and ask for help from you. Trixbox is put on DMZ (Allstream connection); extension are set as follows: Careinvite = Yes ; Nat =Yes ; Qualify = Yes. Trixbox is behind two routers. First one is Cicsco 800A but it is set to public ip, so it's not blocking anything. Second router is a Linksys RV042 (http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2F Layout&cid=1123638171618&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=7161 822279B08" This Linksys router has a specific port for DMZ but it also has DMZ option through of it's other 4 ports. I tried both way and it failed. So, SIP device from outside registers fine. Calls made to any inside extension or outside number dial out fine too. Once the call is picked up then there is no vioce both ways. I tried sending DTMF but that doesn't go through either according to the following log: -- Executing Goto("SIP/4000-09c6e958", "from-pstn|s|1") in new stack -- Goto (from-pstn,s,1) -- Executing Set("SIP/4000-09c6e958", "FROM_DID=s") in new stack -- Executing Goto("SIP/4000-09c6e958", "timeconditions|1|1") in new stack -- Goto (timeconditions,1,1) -- Executing GotoIfTime("SIP/4000-09c6e958", "08:30-17:30|mon-fri|*|*?ivr-2|s|1") in new stack -- Executing Goto("SIP/4000-09c6e958", "ivr-3|s|1") in new stack -- Goto (ivr-3,s,1) -- Executing Set("SIP/4000-09c6e958", "LOOPCOUNT=0") in new stack -- Executing Set("SIP/4000-09c6e958", "__DIR-CONTEXT=default") in new stack -- Executing Set("SIP/4000-09c6e958", "_IVR_CONTEXT_ivr-3=") in new stack -- Executing Set("SIP/4000-09c6e958", "_IVR_CONTEXT=ivr-3") in new stack -- Executing Answer("SIP/4000-09c6e958", "") in new stack -- Executing Wait("SIP/4000-09c6e958", "1") in new stack -- Executing Set("SIP/4000-09c6e958", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing Set("SIP/4000-09c6e958", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/4000-09c6e958", "custom/nightGREETING") in new stack -- Playing 'custom/nightGREETING' (language 'en') -- Executing Hangup("SIP/4000-09c6e958", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/4000-09c6e958' Basically, where there is no voice both ways I hit Hungup but the softphone (SJPhone) takes too long to hangup and when hangs up the following msg is displayed: Critical transaction failed: Client non-INVITE transaction [Trying]: timed out What logs should I look into or what type of tests should I do to find out where the problem is? The same system was tested between rogers and bell connections and it worked fine (except for the Linksys router was not in between) I hope it is not the Linksys router because it is a very new product and it's not one of those old Linksys routers (like WRTG45 and so on....that never worked properly). Asterisk gurus please give me your feedback on this. Thanks in advance, Bruce _____ Get news, entertainment and everything you care about at Live.com. Check it out! <http://www.live.com/getstarted.aspx>
