similar to: System freezes when using Festival with usecache

i have tried to connect my asterisk server to vonage like this: Sip.conf: register => 1<yournumber>:<secret>@atlas-east.vonage.net:5060 [vonage] type=friend username=1<yournumber> secret=<secret> host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1<yournumber> fromdomain=atlas-east.vonage.net canreinvite=no nat=yes

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Greetings all, Recently ran into a seg fault the keeps reoccurring whenever R-java is used (I believe). Not sure if this is an R issue or an openjdk issue so I'm trying to cover all bases with this report. I tried downgrading R version from 3.4.0 to 3.3.3 with the same results. Also tried downgrading java version from 1.8 to 1.7 with no luck. Running on a fully updated CentOS 7 x86_64,

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

We've had this more general topic on R-help, and also in R-devel recently. There's one case here where I get the feeling R never gets into swapping but more directly aborts possibly from a bug we can more easily fix. Today I've been working (successfully! - not yet committed) at fixing str() for very large strings. In this process, I've found that pc <- function(.)

Hi all, I'm not sure if I'm not supposed to do the following (the dyn.unload part, I mean) or this could be a bug (in R or Rcpp): ``` Rcpp::sourceCpp(code=' #include <Rcpp.h> class Object {}; //[[Rcpp::export]] SEXP new_object() { return Rcpp::XPtr<Object>(new Object()); }' ) new_object() dyn.unload(list.files(tempdir(), ".(so|dll)$",

The R script I run daily for hours looks like this: while (!finish) { Sys.sleep(0.1) time = as.integer(format(Sys.time(), "%H%M")) # always crash here if (new.data.timestamp() <= time) next # ... do some jobs for about 2 minutes ... gc() } Basically it waits for new data, which comes in every 10 minutes, and do some jobs, then gc(), then loop again. It

Interestingly enough I had this same problem today.... 1. I created the directory and permissions for the directory " /var/lib/asterisk/festivalcache/ " (per the comment in the festival.conf file) 2. I had to comment out some things in the festival.conf file: the "host" line, the "port" line, and the "festivalcommand" line. I have also noticed the

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

A reply from stackoverflow suggests I might have hit this bug: https://sourceware.org/bugzilla/show_bug.cgi?id=14023 I can confirm that this glibc bug affects my system (latest CentOS 7). However, as far as I know, R is not multithreaded in its core. Is it possible that some library triggered this? Regards, Steve Tomas Kalibera <tomas.kalibera at gmail.com> ?2019?8?2??? ??4:59??? >

Hi , I tried to do the following . Before I execute the R script , i tried to clear the work space using rm(list = ls(all = TRUE)); . I get the following back trace . Its a huge backtrace , attached a part of it . Can anyone help why I get this ? #13 0x00002aaaaafd612f in Rf_eval (e=0x1448d68, rho=0x16c6780) at eval.c:399 #14 0x00002aaaaafdabf6 in Rf_applyClosure (call=0x1447108, op=0x1448f60,

R's dyn.unload() will unconditionally unload the given shared object; it does not check whether there is any object (external pointer or weak reference) with a C finalizer pointing into the space of the shared object being unloaded. So it is expected that R will segfault later when such finalizer is run. Currently there is no other way than to handle this on the side of the shared

Do you think this should be "Bug Reported"? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Samaritoni Sent: Wednesday, October 06, 2004 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Festival, getting gethostbynamefailed error

That would be because the product nc * ni overflows in cbuf = buf = CallocCharBuf(nc * ni); Since we disallow strings with more than 2^31-1 bytes we could test and reject this. It might be more future-proof to change the declaration of int j, ni, nc; to R_xlen_t j, ni, nc; and let the character allocation code reject, but that would create a memory leak since the Free call

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

Asterisk v1.0 is running on RH 9. I installed festival RPM (festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add: (define (tts_textasterisk string mode) "(tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe

In an optimized build, debug info is just an approximation. It might help to debug in a build of R and packages without compiler optimizations (-O0), where the debug information is accurate. However, first I would try to modify the example to trigger more often, or try to find external ways to make it trigger more often (e.g. via gctorture). Then I would try to make the example smaller (not

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried