similar to: cvs asterisk compile failed (newer libpri)

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

I have not used astcc with pin codes so far, since I set-up the phone number as card number. Some of my users want now to dial in to the system and than use their card, which is their phone number. For that I would need a way of authentication, like a pin. I want to use something like: What is your card number: <user keys in the number> Enter your pin: <user enter a long pin>

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

Greetings list, It's been a while since I've been able to focus on asterisk packaging but this weekend I took some time to audit and recompile packages for CentOS 4.2. You can find them here. ftp://ftp.linuxsys.com/ftp/pub/releases/CentOS-4.0 You have your choice of 1.2.1 or 1.0.10 releases. If you need zaptel modules then install this kernel as well:

What has been changed at the ASTCC from the previous head to the current one? How to use the PIN? Can I avoid it? bye Ronald

I try to track down an error that causes that Astcc just reports the time, but not the costs. I could narrow the problem down into this sub routine: sub calccost() { my ($adjconn, $adjcost, $answeredtime, $increment) = @_; eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment }; my $cost; print STDERR "Adjusted time is $adjtime, cost is $adjcost with

The time stamps in ASTCC are useless as they are now: Fri Jul 22 15:06:25 2005 Wouldn't it be better to use something like: 2005-07-22 15:06:24 Fri I want to sort the records by date, but with the format now it is impossible... or do I miss something? bye Ronald Wiplinger

I finished my setup for ASTCC and I am looking for a tool to convert a mysql table to excel and back. Which one is good, and where can I find it? bye Ronald

I have some troubles with ASTCC. TOOOOO often the "in-use" flag remains set. I would like to find a solution, where astcc.agi checks automatically if THIS user is in a call rather than to check the flag. If that is not possible, I would like to have an extension to dial to, and it will after hang up, reset the flag! The in-use flag remains set, if the caller hang up before the

I (cannot sleep and I) am thinking if there is a way to make inwards billing easy possible. To dial out we use something like: exten => _9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF}) (I have an extra field TARIFF, what allows me to use different prices for different users) To dial to a phone we use something like: exten => 888888888,1,Dial(SIP/6001,20,tr)

There is no DNS functionality built into the IAXY, I am assuming you want to be able to home onto a SOHO site with a dynamic IPaddress. I have not been able to figure out a clean way to do theis (make it easy for the roaming sales-folk etc,) who will not have a high tolerance for running something like an IAXYPROV. You would need to somehow have an external fixed address which would

How can I fulfill that? *Billing Increments* Continental USA: six (6) second increments. International: thirty (30) seconds minimum and six (6) seconds thereafter. Mexico: sixty (60) seconds minimum and six (6) seconds thereafter. bye Ronald Wiplinger

I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better!!!! I could use a different context for this customer and use only his account code as card number. Any advice would be

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password, however since it is a cgi script that does not seem to be posible. Any ideas, any suggestions ?

Hi. I am trying to provision my newly purchased IAXy. I tried ./iaxyprov 192.168.0.4 myiaxy.conf It print a bunch of numbers on screen and then say Provisioning is 63 bytes Total packet is 77 bytes Got response back from '192.168..0.4' And that's it. I didnt get nothing else and there isnt any solid yellow light after minutes. I wonder if anyone could explain what I'm seeing? Is

Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback.

Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]