similar to: Aterisk with Realtime

Hi all, I have confgured my oracle application with the Oracle Express Production release database. This all is configured well and tested(I can see that he founds the database). However, when I generate a scaffold application there is no view for the CRUD functions created for the application. Below is listed what is generated for me for a Products table: E:\workdir\ruby\depot>ruby

Hi All, We have just installed Asterisk@Home. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated.

I have a question about the Asterisk Application ICES. I've got Asterisk setup to accept a phone call and call the ICES app which sends it to an Icecast server. exten => 1,1,SetGroup('stream') exten => 1,2,GetGroupCount() exten => 1,3,Ices('contrib/${GROUPCOUNT}-ices.xml') exten => 1,4,Hangup Everything works fine. Unless I have more than 24 phone calls being

Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was relesed. now I am having troubles with my dialing plan and voice mail. As part of the upgrade I re-built the machine so there was a blank slate however after installing 0.7.1 I had no mail box creation script and could not figure out how to go about creating a mailbox, any suggestions would be usefull. I have looked at

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hi Everyone, After a fair amount of faffing ive managed to get the 2000w working with asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls over our BT line). The final solution is to set up outgoing VoIP calls but I now know that without a SIP aware router I can think again! (damn you iptables!) In the mean time I'm trying to figure out why I can't get the

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)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Hi, How does * read and process the extension.conf file?? The reason I ask is that I think it probably has a very large impact on how the calls are routed and processed by the system especially when it comes to least cost routing.. Let me explain...with an example.. I am using the * Devkit to get to grips with the system, so I have and X100P (Zap/1) and and S100U (Zap/2).. Below is my

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

I'm experimenting with asterisk. This is my environment: - Debian sarge (vanilla kernel 2.4.29) - Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g - Two sip phones (One cisco 7905 and one soft-phones X-Lite) - Digi International Datafire Micro V (Europe) (rev 02) (zaphfc) After two days of work now I can call between the two sip phones and I can call from sip phones to outside numbers via ISDN card.

I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.