similar to: How to originate a call from console CLI ?

Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/f14d382f/attachment.htm

Hi! I ran the following test on 3 different setups: #!/usr/bin/php <? $n=1024*256; $usage1=memory_get_usage(); $rusage1=memory_get_usage(true); $a=array(); for($i=0;$i<$n;$i++) $a[]=0; $usage2=memory_get_usage(); $rusage2=memory_get_usage(true); echo ($usage2-$usage1).'/'.($rusage2-$rusage1); ?> ...and I got the following results: 32bit

I want to use Asterisk as a general message delivery system here. That is, I want to be able to have a (shell, perl, etc.) script on my Asterisk server dial an extension, wait for it to be answered and then play a sound file and then hang up, or even wait for a response or reactions to some IVR. Certainly if I had a SIP library, I could have the script simply look like a SIP extension but that

Hi All, I am facing issues restarting the gluster volume. When I start the volume after stopping it, the gluster fails to start the volume Below is the message that I get on CLI /root> gluster volume start _home volume start: _home: failed: Commit failed on localhost. Please check the log file for more details. Logs says that it was unable to start the brick [2013-08-08

Dear all, I am coming across a funny problem in taylor diagram. When I plot the diagram for given values of a variable some points are going outside the diagram. i have never seen such a diagram and hence cannot understand the reason for it. any help in resolving this issue would be greatly appreciated. the taylor diagrams are attached in this mail. thanks and regards Roopa -------------- next

Hi All I have created a group 'test' which has the following users: brajesh_01:/var/log/samba # getent group |grep test test:!:13159:brajesh,yatish,usr1 brajesh_01:/var/log/samba # I mapped this user to ntgroup user, 'Web Master': brajesh_01:/var/log/samba # net groupmap add ntgroup="We Master" unixgroup=test No rid or sid specified, choosing a RID Got RID 27319

Hi All, I am using Samba 3.2.11 on sles 10. I have create a Unix group 'grp1' and mapped it to a ntgroup "Group One". Unix group 'grp1' has one user i.e 'usr1'. 'net groupmap' command show the mapping with correct SID and gid value. Please, have a look here. # cat /etc/group | grep grp1 grp1:!:12613:usr1,raj1,raj2 # # net groupmap list Group One

Hi Floks, This is probably really dumb but here goes: I used to be able to place calls to my SIP phones from the CLI using the 'Dial' command for testing. I have installed asterisk on a new machine and copied over the .confs and started it up. It all works fine. But when I try to initiate a call from the CLI using 'Dial' it just says: *CLI> Dial No such command

Hi All I have Samba 3.2.11 installed on Suse (sles 10) machine. I am playing around the group mapping functionality. When I map a linux group to nt group. I can see that groups is visible on windows client. But, if I restart the smbd daemon (while changing the scurity mode for ads to user or vice versa), I can no more see the mapped group to the the Windows client. Though, I was able to see

I'm trying to figure out why I get delay using the echo test application. I'm using the asterisk cli so I got no external factors that could interfere. I'm getting close to half a second delay speaking into the microphone and hearing it out through my speakers. I'm doing lots of audio work (sequencing and recording) so asterisk is definitely the problem. Of course, only thing

When using call files I see Dial command when on the CLI however when using the AMI I dont see the originate Dial. Is there a logging level that will show the originate dial on the CLI? If there isnt currently a log state for that shouldnt it do that just like the call files do? My logger.conf has: ; debug => debug console => notice,warning,error,dtmf ;console =>

I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/7777 originate SIP/7777 extension 987654321 at outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route is this

Dear all, I am new to plotting Taylor Diagram using plotrix package within R, hence this post. I have written a script which plots Taylor Diagram with one reference and 7 model values. However the font size, line width and line type are not clear when saving the diagram as a jpeg file. I tried the functions lty, lwd and font but no apparent change. I am attaching the script here. Any help would

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):