similar to: Fw: help required configuring card

Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system. what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx. Thanks inadvance regards rudra -------------- next part -------------- An HTML attachment

Hii, I've a Digium TE205P card and I'm running asterisk@home in a box.I want to configure this card in that box.But as it is a dual port card I'm not sure how to configure it in my box.By searching diffrent site I could able to make a sample configuration bellow I'm giving.Basically I want to connect one T1 connection in one port and in anather port of the card I'll

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards

Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox=21 at from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug

Hello, I am new to Asterisk forum :). I have a requirement of terminating 3G Mobile originated calls (coming through 3G-MSC) to EPBX Phones via Asterisk PBX. Setup: Mobile ----> Mobile Switching Center ( 3G)-----SIP interface--->Asterisk PBX--->SIP Phone. I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk server. I know there is a

Hi all, I've setup a PBX in production environment last week but we have immediatly find out drop conversations and lot of errors (in asterisk logs) like: Apr 10 17:30:48 NOTICE[25154] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 My configuration is: Motherboard: SuperMicro P8SCT (Intel E7221 chipset) Hypertrading and APIC enabled CPU: P4 3GHz Digium TE205P

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO modules) and a TE205P dual E1/T1 card. The problem is with the TE205P and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf files. I go in and add the required lines to configure the TE205P and restart Asterisk. All is fine until I reboot the

Hello all, I've got Asterisk and a TE205P. One port on the TE205P talks over E1 ISDN PRI to the outside world (thorugh BT). The other port talks to an Avaya switch, also over E1 ISDN PRI. All is working well, except that when people try to dial out from the switch through Asterisk (with a TE205P) there are no DTMF tones transmitted to Asterisk. DTMF seems to be fine when the

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

I have a TE205P, jumpered for E1, added the missing wct4xxp-line to /etc/modprobe.d/zaptel, zaptel.conf is just span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 Which, according to my reading of the documentation I could find, should be correct. ztcfg doesn't complain about anything, yet all I ever get is RED

Hello, I am working on a project and have a few questions. I want to connect one port of the TE205p to the PSTN, and another port to the PRI port of a PBX. Basically Asterisk will sit between the PSTN and the existing PBX. Are there any gotcha's I need to be aware of? Will the existing PBX be able to dial through asterisk as if it was a PSTN connection? -- Isaac McDonald Got VoIP? imcdona

Hello, Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardware We are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know.

I've been working with Digium, but hopefully someone on-list has an idea... The card will transmit audio to the T1, but everything it receives is completely unintelligible static (in place of received audio only, not all the time). I've verified everything on Digium's troubleshooting checklist and they were unable to find anything when they logged in. zttest yields 100% for the

Hi there, We've got a problem connecting Asterisk with a TE205P to a TeleWest E1 ISDN line in the UK. We get a lot of "HDLC Bad FCS (8) on Primary D-channel" errors, and every so often the Primary D-channel goes down and all the calls got dropped. We've fully tested the card and made sure it's got its own IRQ. I was just wondering if anyone out there has

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

Hello Group, I'd like to hear your thoughts for the following design. To implement a Voice Gateway solution with atleast 2 PRI (60 channel) capacity, is it better to implement - 1 Grunty Server with a BiPRI card (TE205P, 210P) / Quad PRI (TE405P, TE410P) or - 2 Normal Servers (Dell 420/430 with 2 Gig RAM and 80 Gig SATA) with Single PRI (TE110P) or BiPRI (TE205P, 210P) and Load balance