similar to: Processing incoming calls with multiple contextst over PRI

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

Hi experts I have a trouble in access rights I am running Samba 3.0.31 on Solaris 10 x86 64 bits as member server of an Active Directory 2003 R2 domain (MYDOMAIN) using Identity Management for Unix I set rights to access a sub folder of a Samba share. On Solaris the user "toto" jdoe can write a new file. From Windows, the same user can't. Itlooks like OK when the primary group

Hi all, I read the setup at http://www.blackh0le.net/articles/vpn-dun-howto.html to setup my VPN. However, I'm having a problem which I think is proxy-ARP not working. I like to ask you to see if you know what's going on. When I ping 10.77.1.1 from windows XP machine the packets get to the 10.77.1.1 machine, but they don't have a return path to get back. When I do ping the windows

Can somebody tell me where I can find the differences between OPUS 1.0 and OPUS 1.1? ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel]

Hello I googled about my problem and found some hints but not a complete solution. Situation: On laptop 1, I am user oub (uid=1000) and in the sudo list. I connect an USB drive (jfs file system), its get automatically mounted with the correct uid. I copy files with rsync from laptop 1 to the USB. ( rsync -auvz --progress /home/oub/files /media/usb ) However: On laptop 2, I am user oub

In order to put a shared pbx in an office building for multiple businesses, I will have to make sure that the caller ID information going out is correct. i.e. company a's main phone number is 5551212 company b is 5572121 company c is 5596767 Now I know how to distribute incoming calls based on the number being called, but how do you set the caller id going out depending on what

Hello Marc, Am 24.10.2013 21:00, schrieb Marc Muehlfeld: > Hello Thoralf, > > Am 24.10.2013 20:32, schrieb "Th. S?ldenwagner": >> is it possible to hide/restrict access to the home drives of our samba >> users when accessing them directly via netbios address? >> >> The server is running at school and there are several pupils who have >> the ability

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

Hi, I'm currently working on evalutating a AD-Domain for my Department. Since I have a couple of year experince in running a NT-Style Domain, my choice is samba - nowadays AD-DS. Now I'm stuck, and I would really appreciate some more thoughts and a push in the right direction. :-) Thank your in advance Franz The facts: A quick test installation is working as expected - Debian Jessie,

I executed rsync --progress --verbose --stats --recursive test@100.100.100.100::test /home/test2 --stats but each time it showed me files are being transfer but actually the files is not changed or updated any. for example, 12 100% 0.00kB/s 0:00:00 abcdef 44 100% 0.00kB/s 0:00:00 demo2 0 100% 0.00kB/s 157:26:58 demo3 0 100% 0.00kB/s

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

Hi, first post on this list. I run a small game development company specializing in casual Windows/Mac games. We've been using Theora for video playback inside our engine for a while, but we always run into performance issues. I've tracked them down to two parts : YUV to RGB decoding (done via software) and uploading the new pixel data to the video card as a texture. For the YUV to RGB

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus