similar to: module load Crash Asterisk 11.5.1 app_confbridge.c

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

Address 0xfffffffe out of bounds why and how to solve.MyConfbridgeCount(conferencenumber,variablename )return total number of user in conference given by conferencenumber otherwise zero.At runtime using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped.

=================================================================== Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x081b138e in ast_skip_blanks (str=0x0) at /usr/src/asterisk/asterisk- 11.5.1/include/asterisk/strings.h:90 90 AST_INLINE_API( Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.5- 7.el6_0.i686

http://stackoverflow.com/questions/21015596/failed-to-get-160-samples-from-read-factory- asterisk-11-5-1-app-confbridge-c

I have an existing conference bridge running on Asterisk 1.4.2 using MeetMe and it's been pretty much rock solid since it was installed. We do around 460,000 minutes on it monthly and peak at about 150 simultaneous sip channels. I'm adding a second bridge for redundancy purposes into another facility and would like to go with app_confbridge and Solaris. Since it's a relatively new app

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

Hi, I have 2 linux machines on a Gig network. What would be the reason for getting a timeout error. We have --timeout=600 in the command. The version of rsync is 2.6.2. There are no evident network problems otherwise. The command used is rsync -a --delete --timeout=600 --stats --rsh="ssh -p 22" /localdata/ server:/remotedata/ io timeout after 600 seconds - exiting rsync error:

Hello All I've just started to use RExcel instead of CMD Batches. But I don't get the console output to an data.frame and push to Excel. With the CMD batch you got the console out relatively easy with C:\LocalData\R\bin\R CMD BATCH --slave %rscript% %logfile% to a file. But how can I get it to Excel with RExcel an VBA? Thanks, Dario [[alternative HTML version deleted]]

Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading "User Profile Configuration Options" the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty

I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

Hi, I cannot compile Meetme2 on Suse 9.1 and Asterisk rc2. For latter 2 errors I guess I need development for mysql and postgres, but what about first error ? Regards, Robert. In file included from app_meetme2.c:13: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:300: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Dear all I am get the following problem when trying to compile app_meetme2 using mysql...it seems to want to use pgsql.....? anyone my Makefile looks like app_meetme2.o: app_meetme2.c #$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c $(CC) -pipe -I/usr/local/include/mysql -L/usr/local/lib/mysql $(CFLAGS) -c -o app_meetme2.o app_meetme2.c app_meetme2.so: app_meetme2.o

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

Hi, I'm working on a loop function for a large dataset which contains 1000 different groups. I would like to reconstruct the order of events within each group by using a loop function in R. (Currently the order of events are based on the ascending order of prev_event within the group) A demo data frame: event prev_event group 845 0 5360 926 153

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

Dear All, I am using WriteXLS to write tables with multiple sheets with the command: WriteXLS("tables", ExcelFileName = fileName, SheetNames = tableList, perl = "perl", verbose = FALSE, Encoding = c("UTF-8", "latin1"), row.names = TRUE, col.names = TRUE, AdjWidth = TRUE, AutoFilter = FALSE, BoldHeaderRow = FALSE,

So starting up asterisk-11.5.1-2.fc19.x86_64.rpm I get: [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_mgcp.so': /usr/lib64/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_iax2.so':

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of