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...rror code 1 make: Fatal error: Command failed for target `R' Current working directory /tmp/R-1.3.0/src/main *** Error code 1 make: Fatal error: Command failed for target `R' Current working directory /tmp/R-1.3.0/src *** Error code 1 make: Fatal error: Command failed for target `R' -- Govind Vinjamuri NSA Unix Administrator Clinical Research Computing Unit (CRCU) 215-573-5977 -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]s...

...tes for n=2048. Assumption 3: M_PI is just PI, i.e. 3.1415.... Consider n=256, then n2=n>>1=128, n/4=64, n/8=32 first loop: (i=0) T[0] T[1] T[128] T[129] i=63 (last value) T[126] T[127] T[254] T[255] econd loop: i=0 T[256] T[257] i=31 (final value) T[318] T[319] <p><p> -- Govind S Kharbanda Institute for System Level Integration MSc Course Rep. Alba Campus, Livingston, EH54 7EG http://www.sli-institute.ac.uk/~gk/ Tel: 01506 469340 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http...

Hi list, What possibilities are there in asterisk to send an *alphanumeric DTMF*from/to asterisk !? Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e62e7890/attachment.htm>

Hi, I'm trying to make a attended transfer through AMI. I though i could use Atxfer, and it seems ok, but nothing happens. And I can't find any how-to or description on how to do this. What more do I have to do to make this work? In Asterisk Call Manager: Action: Atxfer Channel: SIP/36-xxxxxx Exten: 33 Priority: 1 Context: Phone Response: Success Message: Atxfer successfully queued

...for F77 and CC. I try to configure I get the following message. checking whether f77 appends underscores... yes checking whether f77 and cc agree on int and double... configure: warning: f77 and cc disagree on int and double configure: error: Maybe change CFLAGS or FFLAGS? Thanks in advance -- Govind -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not the subject !) To: r-help-request at stat...

Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110812/84130e1a/attachment.htm>

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

...ost PC for now) so I'm having to transfer the structure back and forth using DMA. So the structure will have to be modified slightly so the values from init->trig are contained as part of the structure. How big should this trig array be? Hope someone can make this less confusing, Gov -- Govind S Kharbanda Institute for System Level Integration MSc Course Rep. Alba Campus, Livingston, EH54 7EG http://www.sli-institute.ac.uk/~gk/ Tel: 01506 469340 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http...

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

...en care of). Source file was keoki_passiton.ogg obtained from the Vorbis.com web site some time ago (I don't think it is there any more). I'm using Version 1.0 Win 32 distribution under Visual C++. The blocks shown above are the first audio block in the file (size 256). Cheers Gov -- Govind S Kharbanda Institute for System Level Integration Hardware Perceptual Audio Decoding Alba Campus, Livingston, EH54 7EG http://www.sli-institute.ac.uk/~gk/ Tel: 01506 469339 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http...

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

Hi.. I'm using gam() to fit a spline model for a data set that has two predictor variables (say A and B). The results indicate that the higher order interaction terms are significant. The R^2 jumps from .5 to .9 when I change the maximum order for the interaction from 10 to 15 (i.e. (AB)^10 to (AB)^15). Is there any way of finding out which of the terms in the model are really

Hi, I ran a small data set from a factorial experiment through R, Minitab and JMP... the result from R is significantly different from what Minitab or JMP give... The data set is at the following link: http://www.personal.psu.edu/nug107/Uploads/2x3_16repsANOVA.txt The first 5 columns are the factors and the next three are responses. In particular, for the response beta11MSE, two of the

...ore) and so also don't know if we're creating the project correctly. If anybody could give us step by step instructions from firing up Microsoft Visual C++ right up to compiling decoder_example.c, we'd really appreciate it as we're very new to this compiler and source code. Cheers Govind --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sen...

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before