similar to: Matching '*'

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

Some questions regarding calling Fast AGI from the dial plan. Considering that the server side of the Fast AGI has to be able to a) use threading and b) connect to MySQL, this causes some serious limitations. I'm not a C programmer, so development options are either perl or python. It appears that the perl DBI isn't thread safe, so that rules out Perl. Python's database interfaces

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

I've noticed that Asterisk AGI scripts don't terminate when a call is answered. Does anyone know how to do this? I would think that this would be a very big problem, if the scripts stayed in memory, doing nothing, until the call terminates. Not only do you have to have a process for routing each call, but all the previous calls, that are still in progress, also have scripts running. It

Well, I'm about ready to throw Asterisk across the room. Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI, if the callee hangs up the call, Asterisk sends a return code, but if the caller hangs up, it does not??? This means if an agi script services a call, and after the two parties have finished speaking, the person who initiated the call hangs up, the agi

Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script. I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton. I've tried this: EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}" and also: SET VARIABLE

Hi. I just want to make sure I understand this before doing something that might break things spectacularly for our users and customers :) We are using Asterisk 1.6.2.9 and my programming language of choice is Perl. I want, when a call comes in on someone's DDI number (which the person who dialled it can only possibly have obtained by dialling 1471 after we called them), to be able to

I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally

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 =>

Folks, When I have a dial string like this: Dial(SIP/3254101&SIP/3254102,20,tr) and I want to check the ${DIALSTATUS} variable after the dial, how do I know which number I am getting the variable for? And, what about this? Dial(SIP/3254101&SIP/3254102@proxy1,20,tr) What happens in that case? How can I get the ${DIALSTATUS} variable for EACH NUMBER dialled? Thanks, Doug.

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: > /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi > install_driver(mysql) failed: Can't load > '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' > for module DBD::mysql: libmysqlclient.so.15:

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm