similar to: getting called number from a zap channel

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Hi folks, I use safe asterisk to startup and run asterisk in the background. In Safe_asterisk script, there is a parameter (right at the top ), CONSOLE which I can set to no or something. If it is no asterisk startup as asterisk -vvvg , if it is set to something the asterisk startup as asterisk -vvvg -c. Now I am running an agi script when calls get hung-up. That is in my extensions.conf I call

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

I have two possible senarios for making a call from an AGI.. Senario1 - Using GoTo In the extensions.conf I have.. [dial-out] exten => _9.,1,AGI(myagi) exten => _9.,2,Dial(SIP/blah/${EXTEN:1}) In the AGI I have.. EXEC GoTo dial-out|9555678|2 So using this method I don't have to really edit the AGI ever if I change the dialplan around as long at the context is correct.. At the end of

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Note: dial string and capi.conf has changed. The main changes are: - added 'relaxdtmf'. - more BSD compatibility - correct PROGRESS handling - added verbose text for capi info/reason error messages. - use correct facility-selector for echo-cancel - added application capicommand() for CAPI

Greetings. I have next feature in features.conf : send => *9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl What it does is parsing CALLERID and DNID from AGI input, performing some actions in MySQL with these values, and then running application for peer (for example, PlayBack) Sounds simple, and it really is. When my user is receiving a call (we are the B leg) and presses *9, everything

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello, We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk Server with Eicon 4BRI card. For most part the service is okay. However, we are are having problems with passing callerID to internal extensions. This is the set of command executed. exten => <pattern>,1,Answer ; Answer the line exten => <pattern>,2,NoOp(${DNIS}) ; debug statements exten =>

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hi Guys I have just installed a call center onto Suse 10. I have managed to do a DBget (astdb) and extract the DNID numbers to play a DNID specific greeting. We have installed Snom 320 and the customer would like us to Send the DNID(nam) to the phone screens so that the agent will be able to answer in the correct language and with the specific customer company name (ie. Agent says

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

I am going to try to figure out why mu asterisk box connected by back to back cable to an PRI appliance is not going to send the PROGRESSING dss1 message. In fact i see the SETUP and the follwing CALL PROCEEDING but not the PROGRESSING so the appliance doesn't allow the "early audio" !!!!! this is my zapata.conf [channels] context = ser switchtype = euroisdn usecallerid = yes

Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf I have in /etc/asterisk/capi.conf 's section "[interfaces]" the following directive context=isdn and the following directive in /etc/asterisk/extensions.conf in

Hi, Im not able to set the outgoing number in filename for asterisk recordings Following is what I have done in /var/lib/asterisk/agi-bin/recordingcheck file . . . include("phpagi.php"); /**********************************************************************************************************************/ $agi = new AGI(); $temp = $agi->get_variable("agi_dnid") ;

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Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

I'm lerning to make my custom configurations. In extensions.conf, there is #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include => from-pstn [from-pstn] include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested