similar to: CAS SF Inband tone signalling problem

Hello, I'm trying to make my asterisk box to act as a telco, in order to reproduce a US environment in europe. Our telco provider is giving us those settings: ESF B8ZF Inbound = E&M Immediate Outbound sig =Wink Start Yield to Glare = Yes Those trunks are using CAS for signaling. I have tried many configs/combinations in zaptel.conf and zapata-channels.conf In zaptel.conf, when having

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 E&M. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 E&M <-> MUX - WAN - MUX <-> Digium TE220, Asterisk I've

Hi, I'm testing an E1 with E&M signaling. Some of the problems I'm running into are the following: 1) if I try to configure any channel above channel 15, I start getting a "multiframe alignment error" on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 e&m=1-15 2) When the test equipment tries to send me

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX========(CAS)===>IVR

hello my friends, when i make a genzaptelconf i get this message ******************** CAS signalling on span 2 conflicts with HDLC with FCS check on channel ******************* Any idea Please? I m installing zaptel 1.4 i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You

Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: > Dear All,

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ;

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve

Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 E&M and PRI. I'm getting an "Integrated T1" which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away.... the

We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward calls to PBX and once that is done, I would try to move the telco

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi CAS is a centralized authentication service, and the cas_auth filter can be installed to automatically make your web application authenticate against a CAS server. The SVN path to the plugin is http://svn.ki.se/rails/plugins/cas_auth and a homepage with some information can be found at http://opensource.ki.se/casauth.html For use, just install the plugin in your application and set

Hi, Could some one explain what are the 4 bits we should define after cas setup (zapata.com) ("CAS Signalling requires idle definition in the form ':xxxx' "? Thanks in advance, Daniel

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

So, I am starting to code this CAS block squashing test and I am wondering if this is something that would more likely extend open Solaris (it does require additional info in zfs metadata that is currently reserved space for future, ie zfs revision++) or if I should code against the FUSE tree where it would could be imported. What type of contribution style is open Solaris? Is zfs

Greetings, I have a customer with an old PBX which cannot accept a PRI. Has anyone tried/tested connecting a CAS T1 to provide 2way DID trunks to a pbx? Either directly to an * server or a gateway? Thanks Jerry

Hi.I'm very sorry for the repost but I forgot the subject. So,? I'm using dovecot 2.0.18 and I'm trying to authenticate through a CAS server (until now authentication was through MS Active Directory). I could not find anywhere some examples, so here is what i have done so far. -install phpcas and pam_cas -edit /etc/pam.d/dovecot ????????????????? auth??? sufficient?????