similar to: auto dialout call status

Hello everyone, I'm using very simple setup to make and receive external ISDN calls through a softphone (x-lite version 3.0 - Win32) via an asterisk box. Hardware setup: - Dialogic Diva BRI (lspci yields: Network controller: Eicon Technology Corporation DIVA Server BRI-2M/-2F (rev 01)) - ISDN BRI line Software setup: - Redhat 9 - asterisk-1.4.16.2 - chan_capi-1.0.2 Asterisk configuration

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>

Hi Guys! I got a Diva passive ISDN card and I can't get it work with asterisk 1.4, It is supported in the kernel as an isdn4linux device but I can't find "Modem" channel type when i type in: core show channeltypes. I'm guessing it is removed in asterisk 1.4. Tried with capi interface but it does not work :( Anybody got some idea how can i make it work or got a link to a

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call.

Hi, I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card. When in Minicom, the only way I can dialout is if i issue ATS18=1 First. Otherwise I get a BUSY message. So thats fine. But when I dialout from asterisk, I get an immediate hangup, so my guess is that asterisk is not issuing ATS18=1 to the ttyI device. Here are my configs, any input would be greatly appriciated.

I am trying to find a way to allow dialout from voicemail when connected from an 'internal' extension context, but prevent dialout when connected from an 'external' extension context. As far as I can tell the dialout context that can be set in voicemail has no regard for the context from which the call to voicemail came in. Any ideas on this? Maybe a variable passed when

I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access to the isdn line? Is there some way to ensure that only Asterisk can listening to the line, or get it to share the line with some other device, such as the fax

I previously posted about this problem and received suggestions involving turning off echo cancellation. As far as I can tell, echo cancellation is already disabled on this channel, so I'm back. What I've got is a small home setup with a single four-port Digium card: Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3:

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

I am trying to understand the difference between the divas4linux available from the Eicom/Dialogic website and the melware version. Am I right in thinking that the melware version is for Trixbox or Asterisk but the Dialogic version is for Asterisk only (i.e. will not work with Trixbox?). I've noticed that some of the things mentioned on the Dialogic web site as included with Divas e.g.

Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear "scratches" and "plops". It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

> > That is possible, how could I check that? > > I can see that IRQ 17 is shared between eth0 and my fritz!card > but I > > don't know if it changes anything: > > Can you try it (or eth0) in a different slot (or change IRQ's in the > BIOS if possible) to see if it makes any difference? That's the > only way > to know for sure, anything else is

Hi, I've a Fritz card which was working fine, recently I changed hardware and my nightmare started. Now when I call someone through the chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I always get hungup. Can someone please give some help? Here are the logs: *CLI> -- CONNECT_IND ID=001 #0x0000 LEN=0049 Controller/PLCI/NCCI = 0x101 CIPValue

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access to the isdn line? Is there some way to ensure that only Asterisk can listening to the line, or get it to share the line with some other device, such as the fax