similar to: realtime caller id extension matching

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

Hi All, I'm testing different ways to implement a LCR/OCN tabe to shift calls to multiple carriers for better rates. I'm using realtime mysql database access, asterisk 1.2.9.1 with mysql 3.23. Scenario 1: I send outgoing calls with with a Goto statement into the context with the realtime switch to dip into the mysql lcr database, currently I have ~13K records/routes. The database

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

Hi, I want to count the number of open Zap channels on my server. [outgoingzap] exten => _0NXXXXXX,1,NoOp(Outgoing Local - 7 digs - ${EXTEN:1}) exten => _0NXXXXXX,2,Set(ZAP01=$[${ZAP01} + 1]|g) exten => _0NXXXXXX,3,Set(UPDATED=true) exten => _0NXXXXXX,4,Dial(${TRUNK}/${EXTEN},60) exten => _0NXXXXXX,6,Busy exten => _0NXXXXXX,7,Playback(thank-you) include => hangupcontext

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

(Both on Asterisk 1.2 and 1.4) I was struggling to find out why my CDR was recording dst = h after a call hangup. It was working fine until I added a GotoIf statement before ResetCDR to calculate some value for userfield column. Today I tested and found out that if ResetCDR is put after GotoIf (or after if in AEL), it doesn't record correct value in dst column, and isntead puts 'h'

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

Hello People, I was wondering if you could take a look at this script, exten => 505,1,dial(iax2/6311${EXTEN},t,25) exten => 505,2,playback(pls-wait-connect-call) exten => 505,3,set(NewCaller=${CALLERID(num)}) exten => 505,4,Set(CALLERID(num)=0${CALLERID(num)}) exten => 505,5,dial(Zap/g1/c/0296389675,20,r) exten => 505,6,Set(CALLERID(num)=${NewCaller}) exten

What's going on here? 555* seems to indicate that the number is being passed as the callerID because NoOp says the phone number. I'm trying to emulate cell phone voicemail where you call your own number to check your voicemail. -- Accepting AUTHENTICATED call from 65.182.165.XXX: > requested format = gsm, > requested prefs = (), > actual format

I am working on this app to dial two external numbers. The second is dialed after the first hangs up. I have simplified things down to: exten => 3852,1,Dial(zap/g1/3964,10,g) exten => 3852,2,Wait(2) exten => 3852,3,Dial(zap/g1/7757,10,g) exten => 3852,4,Hangup Here is the debug: -- Accepting call from '0000000000' to '3852' on channel 0/23, span 1 --

Hi, I'm setting up system with repeated calling of outside extension. When it answers, local extension will ring. Supplied caller id displays correctly on outside phone, but on local extension it's empty. Can I somehow supply proper caller id to local extension too ? Regards, Rob.

Guys I need to send my extension number to a trunk, I'm using Asterisk now, any idea how to do it? Thanks

Dear All, I have a problem with the Marco and the Realtime Extensions in my extensions.conf. The problem is that when I exit from my Marco, I should return to my calling context, which is default but the next step for it should be switch statement which will use realtime extension. Somehow I am getting the following error below with autofallthrough=yes : -- Executing

i have realtime running over oracle database when i have some _ extensions in the database the asterisk won't accept them. Here i tried to call number 47. the extension for this one in the db is: _4[6-9] so the second select should found something with sqlnavigator i find the row but asterisk seems to stop continuing after that i get th emessage invalid extension. May 10 11:58:48

Dear All, Sorry to be posting again. I have solved my problem. The problem is that when exiting from the macro, the priority number is still in effect. For example, priority 1 is at the start before entering macro after the macro the priorty will be 2. Since there isn't any other dialplan command, the switch statement would be search for a priority 2 in the Realtime extensions table. One

Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers.