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...voice packets higher priority. I'm avare of VOIP routers, but they are pricey. Can some of common routers help, or maybe implementing router on another simple Linux box? Any advice, pointers to more info ? How to trace network and debug Asterisk in convenient way ? Thanks in advance, Robert Rozman

...nd the log-rank test) with the formula (Observed-Expected)/(SQR Var). The results are similar but not exact to the M value obtained with the Maxstat. I would like to know whether you are using some correction or adjustment in computing the different ranks. Thank you very much for your help. Dr. C. Rozman Professor of Medicine, Emeritus University of Barcelona, Spain E-mail : rozman@medicina.ub.es [[alternative HTML version deleted]]

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

...NUM}) ;exten => s,3,SetCallerID(${CALLERIDNUM}) exten => s,5,Goto(from-pstn,s,1) and when executed : -- Accepting unauthenticated call from 193.77.90.224, requested format = 2, actual format = 2 -- Executing NoOp("IAX2/guest@252/2", "IAX call from outside "Robert Rozman" <252@posta.etrust.si>: Name: Robert Rozman| Number: 252@postaetrustsi") in new stack -- Executing Wait("IAX2/guest@252/2", "2") in new stack -- Executing SetCIDNum("IAX2/guest@252/2", "1252@postaetrustsi") in new stack -- Executin...

...tracker explicitly said what connection attempts were blocked and why. Check your logs and see. Keep in mind that sip isn't part of the ANY group of protocols. You need to either add it to ANY (not recommended) or set an explicit rule for it. > -----Original Message----- > From: Robert Rozman [mailto:rozman@fri.uni-lj.si] > Sent: Tuesday, December 14, 2004 1:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT- > cancall but cannot receive calls ? > > Hi, > > I hope I won&...

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

...primergy econel server under Suse 9.2 freezes hard... Is this normal behaviour ? Do we have any debug options on loading module ? How to track this problems ? > span=1,1,0,ccs,hdb3 > bchan=1-15,17-31 > dchan=16 > loadzone=fr > defaultzone=fr Thanks in advance, regards, Robert Rozman.

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean && make && make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which

...ople on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a phone that can call then a phone that can autoanswer... > -----Original Message----- > From: Robert Rozman [mailto:rozman@fri.uni-lj.si] > Sent: Tuesday, December 14, 2004 11:24 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT - > cancall but cannot receive calls ? > > Hi, > > I have followin...

Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert.

Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, Robert.

Hi, I'd like to have local extensions accessible through SIP uri (like Joe@company.com), but at the same time for convenince to be also extension with number (like 100) for more convenient dialing thought softphones that support only numeric keys. Can this be done ? Since I'm newbie, I'd really appreciate small example... Thanks in advance, regards, Robert.

Hi, I had music on hold working but now don't know what happened. I get : WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class '') on channel SIP... Any ideas what is wrong ? Regards, Robert.

Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert.

print to standard error output in your perl script: print STDERR "This is how perl-AGI prints to Asterisk CLI output\n"; MATT--- -----Original Message----- From: Robert Rozman [mailto:rozman@fri.uni-lj.si] Sent: Tuesday, August 24, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk Hi, I'm writting some Perl AGI scripts. I got lucky on some working, but try to debug ot...

Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities ("4 for accounting ...). Same happens from SIP or IAX local extension.

Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert.