similar to: Asterisk-Redhat 9 install guide.

I have searched through the archives to no avail. Can someone give a quick overview or starting point for ADSI. I just received a 390 and was told that it is unlocked. But am not sure of the next step... I have tried a limited test using the adsi.conf and received: -- ADSI Available on CPE. Attempting Upload. WARNING[106510]: File app_adsiprog.c, Line 1211 (adsi_process): Invalid or

Peter, I am trying to get memdisk to work with grub and have run into a problem. Grub loads memdisk into the 0x9000 segment and then starts it at location 0x90200. I run into problems in the first printf statement. It turns out that printf uses a switch statement that compiles into a jump table. When I reach this jump table, I believe the code is jumping into hyperspace due to the jump table

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

Hi folks, First off all, I'm new to asterisk. I'd like to make a basic installation of * with a Linejack, as recommended in this mailing list, I downloaded and setup successfuly the pwlib, openh323 lib, asterisk and chan_323. I have made little changes to default configuration files packed with asterisk and I managed to call with the telephone plugged in my Linejack a friend

I sent this yesterday from my wrong email account. yeah, Thank you. I screwed up by only looking at adsi.conf, and not asterisk.adsi. Once I saw that, things started falling into place. Are you able to pass arguments to the phone so that for every call there is unique data on the screen. I see that works for caller id with $Call1p and $Call1s, but what If I obtain an OrderID from my

I am using samba-2.2.3a and samb-2.2.4 in my windows-2000 environment and have the my linux systems configured to use winbind in addition to standard unix for authentication. The problem I am seeing is that linux services and programs that prompt for a password will prompt first for unix authentication and then a second time for winbind authentication. Is there anyway to using winbind and unix

Hi, I am trying to get a prototype working based on Asterisk and Quicknet cards. I currently have to systems set up each with a LineJack card. I have the systems working, but can't get the voicemail demo to work properly. Messages to the user telephone set from the voicemail system are clear, but recordings left through the phone are distorted. Sounds like the audio has lots of echo and

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

>Another option is Aastra's Ring Selectors. They can "split" a line into >multiple lines by ring cadence. > >http://www.aastra.com/Products/RingSelectors/index.html Don't you then have to wait 4 rings b4 an extension will get a * dial ? This devices must wait 2 rings to detect the cadence then * is gonna wait 2 rings trying to get CallerID which I think it will NEVER

OK let’s start out with this. I’m not a pro GUI designer… ? Now that that’s done. Welcome to OpenConf. At least that what we call it now. To config an * file click on the filename to the left. For my example use extension.conf. Now you’ll have a FULL text editor and a parsed list of all the [sections] in the extensions.conf file on your left. On the right you will find any numbered var’s

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Lemme know if you have any patches or add on's are welcome Dave Packham aka p0lar

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right? Dave

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

we are in process of writing a PHP interface for * conf files. we are parsing the files like INI files but the only prob I have so far is that separate extensions in a context dont have any unique tag that I can capture. This works ok [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion

Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P

As part of my work, I am trying to append matrices onto data frames. Naively I assumed that when rbinding a data.frame and matrix, the matrix would be coerced and appended, keeping the names from the data frame. Clearly, I am not fully understanding the process by which rbind works. Example code: > A<-data.frame(1,1,1); names(A)=letters[1:3] ; B<-matrix(0,2,3) > rbind(A,B)

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.