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%2FLayout&cid=1123638171618&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=7161822279B08"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 outWhat 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
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