similar to: 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

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

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

Hello, I have been struggling with the configuration of this card on my box. I have a T1 line and I am trying to setup asterisk with it. I followed all the instructions and I still see a blinking red light on the card. I use fedora 12. If everything is fine should I see a green line when I plug in the T1 line ? I want to isolate the issue so I di not start asterisk. When I run asterisk I get

# Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone = us defaultzone = us Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"

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. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

Chrono here, first I like to say I like your Software and I will do anything to help out on reporting bugs with extreme prejudice, now on with my first bug issue: Computer: Dell XPS M1730 RAM: 4GB Video: Nividia 8700 SLI CPU: Hitachi 32 bit OS: Dual Boot - Ubuntu 8.10 / Windows Vista Ultimate I was trying Skype 3.8 on ubuntu 8.10 and the install was garbage...doesn't even work and I am

I'm Matt, trying to set up my radio station on icecast, I downloaded the ice cast server, what source do I use, I do have shooutcast source? its not connecting? can u help me get streaming? I have shoutcast source, what do I need to do to get connecting? please help, do I need to enter different port number on my shoutcast source, I have teh icecast server and I clcik start runnign it says

I've got an asterisk 1.4.24 box with dahdi complete 2.1.0.3+2.1.0.2 I've got a 2 port T2XXP card attached with on T1 currently plugged in. Inbound calls work fine, but outbound fail with the rather cryptic: [May 19 17:28:07] WARNING[11360]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) There are similar threads I've found on the

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

I am having problem with T1 configuration. Following r the config.. Zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order span = 1,1,0,esf,b8zs # fxsks=1-24 bchan = 1-23 dchan = 24 # Span 1:

Hi I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02), Errors: [Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1 up [Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jan 28 17:32:34] NOTICE[6182]

Hi, I've searched and searched on the possible problems. If anyone can help me that would be great. Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF CRC4 error count: 4750 E-bit error count: 5023 Timing slips: 72 1 TE2/0/1/1 Clear (In use) (SWEC: MG2) 2 TE2/0/1/2 Clear (In use) (SWEC: MG2) 3 TE2/0/1/3 Clear (In

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

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear