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...00:04:14 Dynamic No *52633 0002 N/A -6:-56:-48 Dynamic No 89737 0001 N/A -6:-40:-42 Dynamic No 89932 0002 N/A -6:-39:-20 Dynamic No 65393 0002 N/A -6:-33:-17 Dynamic No * -- Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140122/bdea5147/attachment.html>

----- "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...

...module, or unknown parameter (see dmesg) # cat /etc/dahdi/system.conf loadzone = in defaultzone = in span=1,1,7,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 echocanceller=oslec,1-15,17-31 Is there anything missing or i am going wrong.. Help me out. Thanks in advance... -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100303/12003827/attachment.htm

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20091123/7d894f8e/attachment.htm

...CPU3 CPU4 CPU5 CPU6 CPU7 ... 37: 1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39: 1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140514/0e37def4/attachment.html>

...9; base-bridge <485fcffc-49ad-4e86-8d1b-4655631a232a> -- Channel CBAnn/1701414-00000000;2 left 'softmix' base-bridge <485fcffc-49ad-4e86-8d1b-4655631a232a> -- Channel SIP/8484-00000001 joined 'softmix' base-bridge <485fcffc-49ad-4e86-8d1b-4655631a232a> -- Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141021/b91414f0/attachment.html>

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110110/51355245/attachment.htm>

...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 scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130612/581746dc/attachment.htm>

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

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

// 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

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]:

...tion: meetmemuteall with meetme number. While executing MeetmeMuteAll action, this action will mute all 20 normal users but not admin.. This thing work fine but I want to play "conf-muted" prompt file to all these 20 users simultaneously.. Is there any way to do this...?? -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091216/a0e3d470/attachment.htm

...2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [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 Any Idea..? -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100703/2a7cbcf6/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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110725/1c1feb04/attachment.htm>

...- - - - - - - - - - testlaw - 3000 2 2 - - 1 - - 10000 - - - *8999* - - - - - - But when I call, can't here voice on device where amr codec is loaded. -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120321/af7f83bb/attachment.htm>

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>

...;dundi/cache", keytree=0x0) at db.c:595 #4 0x007e85a2 in process_clearcache (ignore=0x0) at pbx_dundi.c:2204 #5 0x08196e6b in dummy_start (data=0x9a46f70) at utils.c:1004 #6 0x00781a49 in start_thread () from /lib/libpthread.so.0 #7 0x006bdaae in clone () from /lib/libc.so.6 (gdb) quit -- Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140823/6bfa821c/attachment.html>

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 :

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