search for: 4fxo

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To: asterisk-users@lists.digium.com Message-I...

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operatio...

works with Asterisk. I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other equipment that will provide up to 8 fxo ports and connect to asterisk. for future projects I'd also like something with 2 fxo ports and 4 - 5 fxs ports - I suppose a digium card would do fine for 2 fxo and 2fxs and I could do a sipura 2002 for 2 more.

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas...

Hello I have a working * server with a TDM card and 4 FXO ports. We have 4 lines now and need to add 2 more lines (and possibly two more later). I'm wondering the best upgrade path for this situation. The simplest I can invision is adding another TDM400 card with 4 FXO ports, and use 2 now and the remaining 2 later. Are there success stories with using 2 TDM cards? Any info will be

Hi I am using a 4FXO , TDM400P card I am able to call outside , after modifiying extensions.conf with exten => _9X.,1,Dial(ZAP/1/${EXTEN:1}) using this , I can only dial through one of the port , Actually I want to dial outside using round - robin search After reading the manuals , I...

I have an easy question. I setup Asterisk with a TDM400 w/ 4FXO ports and I have the following problem. A call comes in correctly. The callers dials extension 100 (grandstream SIP phone). The caller then hangup, before the call goes to voice mail, however, the phone continues to ring, then goes to voicemail, and leaves an empty vmail message, long after th...

In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL...

My equipement : 2x tdm 400p /4FXO/4FXS 16 PORT ASTRIBANK My first TDM400p is the sync master, so i set astribank to sync to it, but the quality is bad, like it is going fine and second after >>ROBOVOICE ;) any other devices (isdn/sip/tdm cards) are working fine, looks like astribank dont like to be sync 'slave'. Is t...

I have a AEX800 PCI Express card - sort of a TDM800 with PCI-Express. (or AEX844 - 4FXS & 4FXO) I downloaded Asterisk Now - and have got this loaded on a new motherboard (Intel with 3 PCI, 3 PCI Express - etc). (Downside on PCI-Express is the physical support the express slot gives (very little) compared with an 'old' standard PCI slot!!!) With only this card in the box.... "A...

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using cen...

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hello. I am totally new to this PBX/asterisk/telephony world. I understand that what I need between the T100P/E100P and the POTS analog lines is a channel bank/multiplexer? I am looking for recommendations on this type of hardware. I have been looking around and found a product from valiant communications (http://www.valiantcom.com/) and/or

Hello List, I'm from Germany and I want to use a Asterisk System. I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my ISDN-Phone on the Sip-System. My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. The Asterisk has to register is at the SIP Provider and if a Call

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

I've tried this before, with no luck. I've got to try again this evening, and I'm looking for some help. Here's my configuration -- pretty simple, really. Asterisk box -> T100P -> TA750(20FXS/4FXO) -> phones and outgoing lines I have an analog modem (Ok, it's a TIVO) that I need to be able to dial out. Right now, I have the modem connected directly to an outgoing line that asterisk is also connected to. As long as the line's not in use, the modem can pick it up and use it. Wha...

I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to configure asterisk to dial out on the gateway. I have one of the FXO ports configured on sip account 100. If I dial the sip account then the router gives me dial tone, with which I can dial a number. Unfortunately this is not the behaviour I desire. I want to setup the FXO port as a trunk with out it giving me a dial tone. I have

Dears, There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO lines) The outgoing call of the one server may be conflict with the established call of the other one, is any way to force the Asterisk or Dahdi to dial after hearing the Dial tone ? -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists....