similar to: E&M signalling

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

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

Hi All I have problem with LIBMFCR2 for once Exchange I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2. my system: Asterisk CVS 1.1.X LibMFCR2 pre.005 and unicall-pre.005 this's my setting and my setting wanpipe1.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Have anyone tried using this? I've looked at app_rpt, and that's a nice project, but have anyone tried using Asterisk for radio services using a Mux e.g.? I was thinking of using an E&M Mux (or channel bank i think) with TX/RX/BUSY/PTT functionality. Or even tried to decode any signalling commonly used with radio communication?

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

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

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly

Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

Hi all, Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [root@server1 zaptel-1.0.7]# modprobe wctdm ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed

Em (20:59:21), Asterisk Users Mailing List - Non-Commercial Discussion escreveu: >On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote: >> >> Does Somebody know why no load modules to FXS? I used zaptel-1.0.7 >> version. > >What color are the modules? >Post your /etc/zaptel.conf > >-- >Mike Mike, The color is green, red ans bluw. Link

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

I could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

All, One of our customers is using a Telrad PBX, we are providing phone server through asterisk via a T1 using e&m directly connected to the Telrad system. We're using a T1 cross cable as normal, the T1 part works great. No alarms. When we try and dial out the Telrad using a direct trunk group, the call fails. When looking in the asterisk console I noticed only 1 or 2 digits are seen,