similar to: Following completion when Dialing.

As of quite recently I experience failures to compile source packages. Messages like * Installing *source* package 'dse1' ... ** libs g77 -mieee-fp -fPIC -g -O2 -c dsefor.f -o dsefor.o gcc -shared -o dse1.so dsefor.o -L/usr/lib/gcc-lib/i486-linux/3.3.3 -L/usr/lib/gcc-lib/i486-linux/3.3.3/../../.. -lfrtbegin -lg2c-pic -lm -lgcc_s -L/usr/lib/R/bin -lR /usr/bin/ld: no se puede

Hi, there I'm totally new to this. I used to use SPLUS, but think maybe useful to try R at home. But seems it doesn't like me. I got error message says ld: cannot open -lg2c: No such file or directory My OS is openlinux 2.3. I use egcs-2.91.66. I searched under /lib/egcs-2.91.66/ but found no libg2c.a. There's only a libgcc.a. Thought maybe g77 people changed

Working on a Red Hat (not Mandrake) Linux and trying to load my library I get (sorry, can't copy it all, working remotely) gcc -shared -o ..... /user/bin/ld: cannot find -lg2c Does anyone know what the g2c library iss, and if I can download and compile it for Red Hat (I definitely didn't mean to type f2c in the above) Thanks Jonathan

Hello, Here is a problem I have had trying to install locfit, the package for R. I have already contacted Clive Loader who thinks the problem has more to do with R than locfit, so here is the point : The version of R I am currently running is R-base-0.90 that I installed using the R-base-0.90.1-1.i386.rpm package. I dowloaded the locfit package available at the CRAN site and ran : R INSTALL

>From the queues.com file. ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Dear DevelopeRs, I'd like to ask those who develop R packages with compiled code to please try avoiding dependency on GCC (gcc/g77/gfortran/g++) specific features in the code, for the simple reason that there are non-GCC compilers out there that might choke on such features. I found this out back when I was trying to build R-2.3.0-to-be on our dual Opteron based Scyld cluster. Because the

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

I can't compile R-alpha on AMD 64. Rather than include a 1400 line script I have put it on the web http://www.stat.umn.edu/~charlie/typescript.txt way down near the bottom it fails building lapack.so gcc -shared -L/usr/local/lib64 -o lapack.so Lapack.lo -llapack -lblas -lg2c -lm -lgcc_s /usr/lib64/gcc-lib/x86_64-suse-linux/3.3.5/../../../../x86_64-suse-linux/bin/ld:

OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

After tinkering with a hand-knitted extensions.conf based on "Asterisk - TFOT", I've now set up a server with Asterisk@Home and am experimenting with it. I'd appreciated any advice from the more experienced list members about which way to proceed. We (my wife and I) have two separate micro-businesses with two POTS lines plus fax. I'd like to have inbound calls on the two

To: jnf@pcisys.net Subject: Compilation of R under Caldera OpenLinux 2.2 > On Sat, 28 Aug 1999 jnf@pcisys.net wrote: > > > > I am attempting to compile R-0.64.2 under Caldera OpenLinux 2.2. I have > > KDE installed but not gnome. When I ran configure, it correctly > > indicated that gnome is not installed. However when I then ran make, I > > received errors

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

This is a very KGB / NSA / InterPOL / CIA type question, but if I have a recorded file (G.711, no compression) can I feed it into standard in of an application and have it recreate the fax that was send? I don't know enough about the Fax handshaking to understand this. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi how can i retrieve the call unique id of asterisk in the dialplan? I have enabled the cdr logging on a postgres database. In the table cdr each record has a field that assumes an unique id (for example: 1141628669.51) Can i retrieve this from the dialplan? For example: exten => 203,1,Answer exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id}) exten =>

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

Add an Answer() as your first step in your dialplan and see if that help. snip