similar to: Sending ANI to TDM40B FXS?

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

I'm running into an issue where subscribers to our service cannot call certain 1-800 numbers if they have a caller id blocked account (restrictcid=yes). This is on Asterisk 1.0.9 and our clients are using Sipura SPA-2002's. Our provider uses a SIP/PSTN gateway, so we hand off SIP to them from Asterisk. The problem appears to be that when an "Anonymous" call goes out, there is

Hi everyone, Sorry for forwarding and top-posting this email again but its as if my TDM40b has keeled over yesterday. After a few hours last night and swapping the card to another asterisk server (with exactly the same result) I needed to have the FXS ports working ASAP this morning so I have repaced the functionality of the TDM40b with some Grandstream handytones which I already had in

Hi ! I have a TDM40B and i try to use it connected to modem for incoming call data transfert. I have no problem to use it with a phone and a talk communication work fine. But when we try to use with modem, with most modem, we got data carrier for few seconds and channel hungup. < [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1] -- Zap/4-1 is ringing << [ TYPE: Null Frame

I've recently noticed some oddities in my CDR records. In some cases the original CallerID that I've set in the .conf file for the extension showed-up in the CDR as the originating extension (on Zap/ devices on the channel bank), and in other places it was the one that I set using Set(CALLERID(num)=<something>) (SIP/ devices). Digging around a little in the source (and doing some

Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara

Hello All, I'm having problems with international calling via Global Crossing. I'm told I need to send a true ani versus a sudo ani. What is the difference and how can I configure asterisk to do this. Global Crossing is denying calls with sudo anis. Thanks, Keith

I have problem with a TDM40B installation. When i modprobe wcfxs the error i get is the following: /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod

Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their

Hi. I was testing my brand new TDM40B and noticed that when the kernel module is loaded (so the card is powered on), the chip onto the ringer module (the one below the S/N label) is pretty hot (I can't touch it for more than 2 seconds). Is that normal ? Or too much hot could be dangerous ? Matteo Matteo Brancaleoni Espia - Emmegi Sys Admin http://www.espia.it

I have 2 xp100's and one TDM400P. I've plugged a phone into the tdm40b, and when i take it off hook sometimes i get a dialtone, other times i get the message "Power alarm of module2, resetting" spit out to the console from the wcfxs driver does anyone know what this could be? I've tried two different phones, with the same result on both. I changed the signalling to

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

Okay went to: http://wine.dataparty.no/ form there downloaded the rpm, http://wine.dataparty.no/rpm/unstripped/wine-cvs-unstripped-013001-1.i386.rpm downloaded the config file and worked through the install guide. I edited the config file to mirror my window setup: c:\ = \mnt\c d:\ = \temp e:\ = \web f:\ = \documents g:\ = \music h:\ = \mnt\cdrom i:\ = \mnt\zip then as a check did $wine -v and

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Yes its a T3 split into seven trunk groups with one D channel and NFAS on each. Can you explain the cut function or point me somewhere please? Thanks, Steve -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or

I've got an interesting problem. I just recently upgraded an asterisk server from a may 9th cvs to a 9/29 cvs. When I did the upgrade I was unable to place calls through the PRI. The calls would process fine, but the provider would reject the call and send me a cause 69 or 77 failure code "Unallocated (unassigned) number". Upon contacting the provider, they told me I was

Morning, I've installed asterisk on FC4 and had a few problems with zaptel stuff. Having installed it on SuSE I was able to check a few things that were different. When installed with the RPMS, the udev stuff gets put into the normal 50-udev file under /etc/udev. On SuSE, it worked when the necessary zaptel section was in a separate file name 55-zaptel.udev (the name is arbitrary,

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access