similar to: 5 Channel / Trunk ??

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and "Advantex" the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the

Dear Group, I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX. On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality. Does anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving

Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf

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

Is it possible to control the dialing format used when making a call from asterisk separately from the inbound format? In my case I have a single robbed-bit T1 configured for featd (to receive ANI and DNIS) on a TE410P. This configuration works fine for inbound calls, however when an outbound call is made on this same T1 asterisk dials outbound in the featd format. I need asterisk to only dial

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

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 am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

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

Hello, I have a T1 (not PRI) plugged into my Asterisk server. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. Thanks. Here are snippets from the relevant files: --

does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set("Zap/49-1", "DID=1231234*4812*") in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important

Hi, Ok, I put the new TE410 into our system last night - This is the second TE410P card. There is also TDM400P in Slot 5 with IRQ 9 which is shared with aic7xxx (SCSI) According to "cat /proc/interrupts", the new card is in the second slot using IRQ 11 (the first card is in slot 1 on IRQ 10 - but always has been). With only the original card install, there has been no issues with

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) ;

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is

The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to

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

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me: