similar to: place T1 calls and ignore/override call progress

Is there anyway a T1/PRI can ignore the ANSWERED signal and just go straight from a dial command to the call was answered? I have a PBX that when calling a certain analog trunk it is not giving me signaling that the call was answered however I hear the PA system come off hook and give dial tone. So is there something in the dial command that can override the signaling and just go straight to

I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry

I have an older Newbridge Mainstreet 3624 upon which I'm terminating some analog DID lines. They are effectively loop-start trunks with battery supplied by me (ie. FXO) and consumed by the serving central office. One major part of DID is the requirement for providing Answer Supervision in the form of battery polarity reversal on the analog trunks. Without it wierd things start happening, like

I'm ordering a PRI T1 for use with Asterisk and a Digium Wildcard TE405P. The provider is asking me a number of questions about how I want to configure the line. Here goes: 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, E&M, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start -

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

>>I'm not sure that your question is clear. You'll probably want to be more specific. >> What is pulse? You mention "as a user", are you talking about voicepulse.com ? >> What are you trying to do with pulse? >> What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as

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:

> -----Original Message----- > From: Jerry Geis [mailto:geisj@pagestation.com] > Sent: September 8, 2004 2:19 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] T100P calls with playback starts speaking > before pickup > > > Hi > > I am using a T100P connected to a panasonic phone switch using T1 and the > switch has 4 analog lines

Just some clarification here. On Monday, June 30, 2003 12:04 AM, Steven Critchfield [SMTP:critch@basesys.com] wrote: > > As for slips and bipolar violations... > T1s are just high speed serial lines. A sleep is when you loose sync > with the far side and when you see a 1 come across the line, you may > not > know which bit it was for. A slip is when one end of the circuit 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:

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {

We are trying to debug a connection between Asterisk and a legacy PBX (Mitel SX200). We turned on the Zaptel debugging and we get the following message quite frequently: Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:09 localhost last message repeated 3 times -- Oct 12 07:14:11 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:11 localhost

I am looking for some great instructions on using asterisk with pulse. I'm using centos 7 and pulse as a user and not having much luck. I have changed all permissions for the asterisk directories. set asterisk.conf user and group to be my user that is running. No go. Anyone done this? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

Could anyone with any experience with * over a non-PRI T1 help this newbie? I have a fractional T1 that is working fine through a channel bank, but I can't get any response on * using a T400P. My analog line and the extensions work fine, but the T400P port 1 shows a red led (what exactly does that tell me?). I brought up zttool which shows a condition red also, but ztcfg shows nothing amiss.

I will second that... I have been doing dedicated IP service for my customers for $130/month in Seattle + loop. (most loops are add about $200-300/month). Anything higher is really a rip-off. John :) -----Urspr?ngliche Nachricht----- Von: Huddleston, Robert [mailto:RHuddleston@cavtel.com] Gesendet: Tuesday, June 14, 2005 12:49 PM An: 'Asterisk Users Mailing List - Non-Commercial

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> I recently upgraded my Asterisk system to Dahdi and now I have an echo problem.<br> <br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a HARDWARE echo

Can anyone offer up a suggestion on a reliable and cost effective customer premise hardware setup to be able to take an inbound IP T1 and deliver a PRI interface to a remote office? Trying to reduce the amount of hardware required to implement this, right now we use a Cisco router to take the IP T1 in on a serial port and then we go Ethernet to a slimmed asterisk box with a single port T1

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate