similar to: When does local leg in call file start?

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

Greetings, This may be a dumb question, but here goes. When I was on 1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to line supervision on my POTS lines using a TDM400P/TDM410P. Since upgrading to the DAHDI branches of 1.4 (SVN and 1.4.26.1), I've only been able to duplicate the success of the 1.4.21 functionality once. To test what I'm talking

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 #<< ==== I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

Hello there, I'm using a Debian box with Asterisk 1.8.13.1 as a DID-PSTN gateway, so I can receive calls in a DID number and redirect it to my mobile line. It has been working flawlessly for a few months, but I have noticed that some calls were not cut after losing one leg (the one with the DID server), and kept the PSTN leg active until I restarted the server (with the unexpected cost

Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and

After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yuan Liu

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the

Hello Guys, I am trying to make Asterisk do dial-out calls. It doesn't even do test calls. It never calls. I tested everything and i am clueless. However i can call Asterisk and it picks up the phone and executes the dial-plan. However, my dial-plan is supposed to do outbound calls. Zap is configured correctly. I am using a TDM400 card from Digium with 4 Fxo ports and i have

I know I've seen this reported already, and I can't remember the fix. I have two ATA186s talking to an asterisk server. When I call in on an outside line, both ring, and I can pick up either and talk. But if I try to call from one of them to the other, the remote end rings just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a