similar to: Hwo to get CallerID: SIP -> ISDN

Hi! As reported earlier this week, I have problems with a sometimes-crashing Asterisk. In most of the cases safe_asterisk is able to restart it. But sometimes it crashes, so that manual interaction is necessary. The seg-faults and crashes occurs, right after call between a SIP Terminal and a legacy PSTN Terminal (PRI/Euro-ISDN), but only if the SIP Terminal hangs up as first. No problem, if the

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

Dear LLVM crew, I have been writing a new LLVM backend with a very simple instruction set. moving a constant value to a register is selected as a load of the constant, with a PC-relative positive offset. Thus, the constant should be stored in .text section, after the load instruction, and offset should be computed at compile time (not link time). I struggle to find out : - how llvm handle

> De : Joerg Sonnenberger [mailto:joerg at britannica.bec.de] > > Well, MCFixup is what a relocation is represented as internally. They get > resolved to a fixed value and replaced, if evaluateAsAbsolute is true. > Hi Joerg, Actually it fails to get resolved to a fixed value. When I compile to .o then objdump it I can see that a symbol holding the constant value was created in

hi R users I have a data set with the name AA AA<-1:100 Now I want to get a child array from AA every 10 numbers e.g. ab =c(10,20,30,40,50,60,70,80,100) How could I subtract aa from AA? thank you . -- TANG Jie Email: totangjie@gmail.com Tel: 0086-2154896104 Shanghai Typhoon Institute,China [[alternative HTML version deleted]]

> De : Joerg Sonnenberger [mailto:joerg at britannica.bec.de] > > The basic idea is to still create a relocation on the MC level, but fix it up later. > You didn't say if your ISA is fixed or variable length, the former, can simplify > this quite bit. The constant island pass generally tries to address two issues: > (1) Merging of identical constants. > (2) Splitting the

dear all, how can i get a vector that shows the number of the column of matrix that contains the maximum of the row ?? can?t believe in need a loop for this... i have a 100 x 3 matrix and want to get a 100 x 1 vector with values 1,2,3 . there must be a simple solution. i just cannot find it. i think am searching on the wrong end. thx for help in advance. m.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everybody, I would like to use managesieve protocol and cmusieve plugin at the same time. I have used the cmusieve plugin for years and I now also enabled the managesieve system. I got the managesieve system working but when I activate the managesieve filter for the user the cmusieve sieve filter is not used anymore. How can I have both sieve

I have > df quantity branch client date name 1 10 1 1 2010-01-01 one 2 20 2 1 2010-01-01 one 3 30 3 2 2010-01-01 two 4 15 4 1 2010-01-01 one 5 10 5 2 2010-01-01 two 6 20 6 3 2010-01-01 three 7 1000 1 1 2011-01-01 one 8 2000 2 1 2011-01-01

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

Hi! I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with backports.org). The HW I am using is Digium's E100P on an HP DL 380. Quite often it crashes, e.g. after a call has finished. Below some logs form the * Console as well as from the /var/log/asterisk/messages (Replaced some stuff with XXX). Any idea what there could be the reason for this segmentaion fault? What other

Hello Lady or Sir, I want to do model selection with 'step' function. But there is a trouble to me. For example, there are 5 independent variables, X0,X1,X2,X3 and X4; and 1 dependent variable y. I want to run stepwise regression y ~ X0 + X1 + X2 + X3 + X4, and I want to keep X0 always in the model. Every time when I run command 'step' with 'keep', error information

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

Hi, I'm installing E100P for isdn pri line. My configuration are like this. zaptel.conf ======================================= span=1,0,0,ccs,hdb3,crc4 loadzone = us defaultzone=us bchan=1-15 dchan=16 bchan=17-31 zapata.conf ================================================= [channels] context=default switchtype=euroisdn context=default signalling=pri_net usecallerid=yes

Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel

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

Hi List, i've set up an Asterisk 1.6.0.3 Server equipped with an Xorcom Astribank BRI XR0013 (2 Port BRI) and Dahdi 2.1.0. In- and outgoing calls are no problem. But I can't get Asterisk/DAHDI to use the second Number of the BRI. All calls set the primary number as callerid. It wouldn't be a problem, if we would use this one as id for our company, but it's already in use (private).

hi, I have a problem with passing caller id information from telco (isdn) to sip client (grandstream). i see callerid in asterisk verbose console but on grandstream (sip) phone is just internal (own-gs) 101 number. Isdn line is connected with hfc card and p2p , asterisk is latest CVS in extensions.conf i have: exten => 2442242,1,Dial,SIP/101,r|T and in console is this: -- Executing

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

I've a strange problem with CallerID when calling Vodafone mobile's from my Asterisk Box. If I dial out on my ISDN-30, setting the CallerID to my DDI (XXX802), a Vodafone mobile correctly shows the incoming call with this number and if the phone's not answered it shows a missed call from XXX802. All very good. However, if the call is not answered and goes to the Vodafone messaging