similar to: replacing step-by-step giving echo

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hi list, we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). People from telco told that they can't connect two PBX's using E1/T1 or only with QSig signaling. I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 would be used. The PBX has a spare ADQ card installed on which we would connect. Has someone a such working setup? Is it working well? More

Hello everyone. I want to migrate an old PBX which uses the ?1-PRI from one Telecom to VoIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is ?1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to connect a PC with 2

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

Hi, all ! I have a situation like this: [SIP Terminals] <-> [*] < -ISDN-PRI-> [Panasonic D500] <-> Telecom (conn to Telecom is with second PRI card in Panasonic and 16 POTS lines). Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16 POTS lines to telco and 32 (advanced hybrid telephone type) extensions. Idea is to have possibility to have some users on

Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. Thanks Sam

My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or

Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. the most important If we want to

Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks & Regards, Vidura Senadeera, Network Engineer,

Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Folks, I really hope you can help me here - I'm beginning to tear my hair out! About 10 days ago my company moved to a new office. As a result of this, we've plugged our PBX box, which has happily been running for the last three years, into our new E1 line. Since then, I've been seeing intermittent yellow alarms. Obviously, since this was working fine in the old office, the thing to

Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there

hi folks. does any one have experience setting up E1 PRI in Shanghai, China? it works fine when we use SIP phone to dial out, however when using forward function on the same phone, it seems like it's dialing out but there's actually no respond from the phone company (China Telecom) and eventually the dial command will timed out. here's our PRI portion of zapata.conf:

We deployed a PABX in China, orginally it used Netcom????'s E1, the zaptel.conf is as following: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 loadzone=cn defaultzone=cn However, recently customer changed to use China Telecom??????'s E1, it always show REC, RED/REC, RED, cycling alarm when I run zttool in console. They sometimes still can make call, but the quality was quite

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. I have the following components in my call routing schema: - PBX with E1 port. - asterisk router with TE405P card(32bit/4 E1 ports). - Teles server with PRI interface card(3 E1 ports) and VTM

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Hi guys, just to let u known the pinout for the adapter : Adapter for connect the E1 telco lines on my digium card DB9 RJ45 3 1 8 2 2 4 6 5 Adapter used for connect the Hicom150 traditional PBX on my digium card (cross connection) DB9 RJ45 3 4 8 5 2 1

Hi all, I am having this problems for a while and could not figure out the cause of this. I have FreePBX version of Asterisk (1.8.11-cert) routing calls to 10 different FreePBX servers (same version of Asterisk) depending on the destination numbers. The incoming calls into the main Asterisk server with 4 x Sangoma A102 E1 card are coming through a SS7 link from an ISUP interface of a