similar to: use of .exports file in asterisk

----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short

Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using "meetme list" command it shows Minus in activity column. Any Idea. >meetme list Conf Num Parties Marked Activity Creation Locked 54682 0002 N/A

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

Hello All, I am trying to install Asterisk 1.8.13.0 & dahdi-complete 2.5.1 & libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. I am using "SunOS solaris-server 5.11 11.1 i86pc i386 i86pc" on Virtual Box. -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was

Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts

Hi All, I have followed below steps to enable echo cancellation. # cd /usr/src # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 # tar xjf linux-2.6.28.tar.bz2 # tar zxvf dahdi-linux-2.1.0.4.tar.gz # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi # mkdir /usr/src/dahdi/drivers/staging # cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging # sed -i

Hi All. I have export some db parameter in /etc/bashrc as follows ... export DB_NAME=xyz export DB_IP=1x.1x.1x.1x export DB_PWD=dkjfaoi Now, I want use these all environment variable into /etc/asterisk/res_mysql.conf file. Is there any way to do this..?? -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I would like to configure AMR codec in asterisk 1.8.9.1. After lots of research i found " http://sourceforge.net/projects/asterisk-amr/files/" thie link, and follow steps to configure amr. codec_amr.so successfully compiled and load in asterisk. *> core show translation * Translation times between formats (in microseconds) for one second of data Source

Hi I have created meetme with 3 user. When i going to mute user it gives following error.. *Asterisk Version : 1.6.2.6* -- <SIP/52987-00000040> Playing 'conf-muted.gsm' (language 'en') [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul

Hi All, Anybody have idea that how to add VP8 codec into Asterisk 1.8 and from where to download. Please share if anybody has idea or related document. -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120502/b5cad0e9/attachment.htm>

Hi, I have tried to start asterisk 1.8.13.0 using "asterisk -vvvvvvvvvvvvvgc" and service asterisk start. Every time I found below kinds of error. Please help me out, if anyone have idea Reading symbols from /usr/lib/libpq.so.5...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpq.so.5 Reading symbols from /lib/libldap_r-2.4.so.2...(no debugging symbols

Hi I am using asterisk 1.6.0.5. I have one conference say 1234786 and in this conference 25 users are talking with each other.. In this 25 users, 5 is admin/marked and 20 are normal.. Admin user has rights to mute/unmute all user by executing action: meetmemuteall with meetme number. While executing MeetmeMuteAll action, this action will mute all 20 normal users but not admin.. This thing work

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer

hello, Im still working on ruby koans. Now I have to do some sandwhich code. The exercise looks like this : require File.expand_path(File.dirname(__FILE__) + ''/edgecase'') class AboutSandwichCode < EdgeCase::Koan def count_lines(file_name) file = open(file_name) count = 0 while line = file.gets count += 1 end count ensure file.close if

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hello, I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir. Also, I can find app_meetme change-logs on the asterisk website. However, the dialplan doesn't have this cmd. I have checked menuselect but it says it has been replaced by app_confbridge. Also, If that *is* the case, does ConfBridge (the newer version of meetme)

https://bugs.freedesktop.org/show_bug.cgi?id=70511 Priority: medium Bug ID: 70511 Assignee: nouveau at lists.freedesktop.org Summary: nouveau_bo_name_get segmentation fault while running root tutorials/gl/glbox.C QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified

Hi! OS: CentOS Linux release 7.1.1503 (Core) Dovecot: 2.2.18 Run user listing - doveadm user '*', single user - work. auth[20540]: segfault at 8 ip 00007fd554340c6b sp 00007ffc17c886e0 error 4 in libauthdb_ldap.so[7fd55433b000+a000] dovecot: auth: Fatal: master: service(auth): child 20540 killed with signal 11 (core dumped) dovecot: auth: Error: net_connect_unix(auth-worker) failed: