similar to: connecting Avaya Partnet with asterisk , TE205P

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 | |

----- Original Message ----- From: rnacharya To: asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 5:56 AM Subject: help required configuring card 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.Basically I want to connect one

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

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 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

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

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

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

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 after some testing with A@H, i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and

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

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

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

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can be shown as avaya--PRI-asterisk--h323-avaya When I do call as avaya user I see name of remote end avay user, i.e. connected line info. As I see in debug remote side send is as 14:07:29:758 Received H.2250 Message = { 14:07:29:758

Dear mailing list I have a data that have 20,000 rows and 20 columns. Io wonted to extract the 10th row only. Example the 10th, 20th, 30th 40th…..20000 th. can you please help me how do I do that.Than kyou. Example is below. Inpute: AG GG GG AG CC CC CC CC CT CC CT CT GG GG GG GG CC CC CC CC GG GG GG GG CC CC CC CC GG CG CG GG GG GG GG GG *CC CC CC CC* AA AG AG AA AA AA AA AA GG AG AG GG GG AG AG