similar to: RBS T1 DID issue

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

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

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

Hi All... Well, I'm still experiencing LOTS of dropped calls since installing the new (non pri) T1 here... I keep noticing a few things in the logs when this happens, namely the "Wink/Flash" statements and the "Didn't get a frame" messages... Anyone got any ideas on if this is a telco issue, a wiring issue, or an asterisk issue? Been trying to track this down via all 3

Hi all, I am seeing on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 (repeated much more then what I will show here). I see that it comes from static void* do_monitor(void *data) in chan_zap.c, but I do not understand what does it

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

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

Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of "sip show peers") which normally shows me when a peer is having connectivity issues. Several that I have

Asterisk is getting red alarms on my T1, sometimes once or twice a day, but today it happened 5 times. Even once is too many. Every call in progress is dropped. Please help! What do I need to do? What can I try? I've googled and searched this list and can't find anything. Here's an example from the logs: Feb 7 13:37:54 WARNING[32451] chan_zap.c: Detected alarm on channel 6: Red

Has anyone here used T1s with RBS with asterisk? Cary Fitch

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

I currrently have Asterisk 1.2.8 with a TE110P Zaptel card tied to our Nortal DMS-10 switch via T1. This T1 trunk is configured for Feature Group D MF. The purpose of this is for LD delivery to a VoIP LD provider as a LD choice for our customers off of the DMS-10. When I sieze a channel on the zaptel, I get a spill of digits from the switch that is the called number, and I can see that in

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

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

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my