similar to: Asterisk 'n Dahdi on Sun Solaris

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

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

----- "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

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 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:

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.

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, 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 have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error; make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]:

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 am using Asterisk 1.8.5.0. In which, found file_name.exports file. I would like to know what is this file contain and what's use of this file.. Like say application app_meetme.c have app_meetme.exports Can you give brief or detail idea about same. Thank you -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

While playing with DB function in Dialplan, I have added some garbage in AstDB. These are some family names with space in them. See this, demo*CLI> database show /18-05-2011 00:00:0052011175221575/TESTDATE : 2011-05-14 21:33:46 /18-05-2011 00:00:0052011175221575/TEST1 : 410 /18-05-2011 00:00:0052011175221575/TEST2 : 155 /18-05-2011 00:00:0052011182614252/TEST3 :

Dear all, I know this is not imq mailing list. But many of the users over here have done exactly what i want. Requirement:- I want to tight bound eth1 for 100 kbps but after i want to create many classes of 64 kbps or 50 kbps and others. But the total sum of classes is more than 100 kbps so my eth1 is not restrciting total bandwidth at 100kbps. According to search on google imq is the solution.

I am running my Mongrel Web Server 1.1.5 on Sun Solaris 10 server with Apache 2.2 proxy running with single Mongrel instance to handle my RoR Web Applications request. My RoR web application was developped on Windows plateform, once I svn my RoR souce codes to Sun Solaris Web Application server and run the application through web browser. My Mongrel Web server most time may die and a core dump

Hi thanks for your posts so far. Since what I had assumed was a 32 bit kernel which I compiled didn''t work, I religiously set out on compiling a 64 bit version as my system is currently booted in 64 bit mode. I installed SUNStudio12 on the system and set the SPRO_VROOT to point towards it and when I do a make all in usr/src directory, I get the following error when make cwds and runs in

Short:     Which package do I need to install to get "auto/I18N/Langinfo/autosplit.ix"? Long: The problem I have was discussed almost a year ago (see this thread), but the resolution was not complete... I''m using Mark Johnson''s slim.py script (package list below), against repo=http://pkg.opensolaris.org/dev, followed by `pkg install xvm-gui` and `svcadm enable