similar to: What hardware to use for ISDN in Romania

Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP!

Hi, I have a strange problem on some isdn modem channels. (* 1.0.9 / chan_modem / 2xHFC-S cards). Everything works fine but when the 2nd (and 3rd etc..) call comes in and * answers and there is about a 1/2 second of sound from the previous call (ivr) before the sound from the new call is heard. It just sounds bad and is quite annoying. I am assuming this is sound that is still in a buffer

> Maybe there isn't a glibc version (although > it shouldn't be hard to port the Solaris version to >glibc). I will try to port nss_wins to glibc. I made some changes to smb.conf and follow the tests described in Samba diagnosis (www.samba.org/samba/docs/diagnosis.html) smb.conf [global] workgroup = NETMANAGER server string = Deltacom Samba Server printcap name =

Quick question: Is there a way to distinguish between calling MSN's when using chan_misdn? More info: I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base number plus 5 MSN's. Now I want to my * to do different things when receiving a call on from different MSN's (like forwarding the call to my FAX machine or forwarding the call to my mobile). The obvious way

Hello; we are looking to replace our current PBX with a *-box; it is connected to ONE ppp isdn connection that is terminated by the NC. We got on this box 4 msn's configured. currently we are working with pstn fxo's behind the PBX; it works but we can't use the CSID information behind it. We want to migrate and keep the MSN's to decide routing in combination with the CID.

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

I have successfully placed a call into and out of the card from Asterisk using vISDN. The current vISDN snapshot now contains the PCI id's for the card so no patching should be required. Most of the initial testing failure was due to misconfiguration of vISDN by me and a bad entry in my sip.conf resulting in one way audio. So I have established that at least 1 channel of 1 port works in TE

Hi All, Could somebody please help me to understand the following: - We have 8 msn's 383590, 383591 etc. What I would like to do is for the person at extension 1 dial out on 383 590, the person at extension 2 dial out on 383 591 etc. I have got myself so confused that I need major help!!! If you could give me a simplistic example, including which files I put the coding in (i.e.

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

OOPS <blush> http://vcalinus.gemenii.ro/quaggarealms.html is what should have been posted by me... This deals with the metropolitan / other issues, I think. I can''t read it... -- gypsy

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hi Guys, We are live streaming the Dojo sessions today ( thanks to hostway for getting this up ). The live streams will be at http://www.centos.org/media.html once the sessinos start. The scheduled kick off is 10am localtime ( UTC +3 ) - KB -- Karanbir Singh +44-207-0999389 | http://www.karan.org/ | twitter.com/kbsingh GnuPG Key : http://www.karan.org/publickey.asc

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then

I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814

Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however..... What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf

Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local

Hi, My name is Monenci Nicolae. I'm from Romania, an i have a problem that i could not find any answer on net. I have an "Samba version 3.013" instaled. I have atach-ed the smb.conf that i have configured my self. So my problem is: I have a network of 10 PC. (5 windows XP, 2 windows 2000, 3 windows NT) I have installed 1 new PC with windows XP. After i installed

Hi, I am using a Netjet-s ISDN Card, and am having some trouble dialling out (Incoming Works Fine). TRUNK=Modem/ttyI0 exten => _90XXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _90XXXXXXXXX,2,Congestion I get the following when diallingout: -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Modem/ttyI0/04XXXXXXXX||Ttm") in new

Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver.... The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit-