similar to: Asterisk with T1

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD

Hi, This may sound crazy but I will go ahead anyway. Here's my scenario: ifconfig bond0 10.10.10.1 netmask 255.0.0.0 ifenslave bond0 eth0 ifenslave bond0 eth1 brctl addbr stp0 brctl addif stp0 bond0 brctl stp stp0 on Is this a valid scenario? I am trying to bond two interfaces and run STP over the bonded interface..The problem I am trying to solve is to run STP over aggregated links

Following the successful deployment of asterisk servers at several of our branch offices, in the near future, I'll likely be implmenting an asterisk server at our HQ. We currently have a T1 PRI terminated on a legacy PBX. I'll be doing a stepped deployment in which, via a dual T1 linecard, the asterisk server will initially pass all incoming/outgoing calls directly through to the PBX.

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002

Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To

Sadly it's a limited to PRIs(at least that's what I've been told) You just can't send that much call data with a good old non-PRI T1. The up side is you get one extra voice channel to use as compared to a PRI. It's strange that the carrier doesn't have PRI cabability, I've run into many more carriers that can't do non-PRI. I've even had one that couldn't

Hi, How difficult is to setup and maintain an Asterisk PBX with phones from multiple vendors? Is it even worth considering or is it safer to pick one vendor for phones and stick with them? I am more concerned about proprietary DHCP extensions, firmware upgrades etc..If anyone has any thoughts or experiences they would like to share I would be more than happy to hear from them. Thanks -Ravi

Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is

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

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

Hello All, How do I specify a variable that has a space in it? I''m trying to setup a yum repo that has two different baseurl''s based on the product name ($productname). Default goes one way, but if the product is "PowerEdge 2650" then is a different URL. The thing is, that is only picking up the default. How should I specify the value with the space in the

This is a follow up to an earlier post. Looking for a means to "individualize" incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to "enter" the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into

Are you looking for local or Long distance? What will these T1s primarily be used for(inbound/outbound, domestic, local, long distance, international) How important are per minute rates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're

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

Hello Everyone, I am new to asterisk and linux too, I managed to installed asterisk on redhat8 with the help of mailing list archives and Handbook guide. I configured 2 SIP phones (grandstream) and it is working fine internally. We have T1 Line coming in with block of 200 DID Numbers. I want to assign DID Numbers to each of SIP phones as an extensions and able to call any PSTN line.

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XXXXXXXXXX for s at test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode

Does anyone know where i can get DIDs in New Zealand. I am look for area code 06. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004