similar to: Meetme Show Activity in Minus

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

----- "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 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 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 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 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]:

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.

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

Test ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I am using four models user, member, unit, society I have linked member, unit and society through member_property table user has one to one association with member while all other models have many to many association. How do I link user to member, and user to unit through member i wanted something like <%= @user.member.unit.unit_number %></p> Can someone guide how to

Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i

// Please reply this email to jai-Oc46A42Di8jtC6uA/Zb986xOck334EZe@public.gmane.org // Hi, My name is Jai, I am a Technical Recruiter with International Business Solutions. Ltd, NJ. My client in NJ is looking for a JD Edwards Technical Consultant. I have included a description of the project below. Please take a few moments to review, and let me know if this is something you might be interested

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: