similar to: Unable to play dialtone on channel xx (Zaptel TE405P)

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI Christian Hoffmeyer YottaDot

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Hello all, My set up is a fractionated t1, with 1-6 voice channels, 21-24 data channels. I have a t100P installed in a amd 1500 with 512meg. Question: In order to set up the data channels, is the following correct? 1) compile new kernel with generic hdlc support 2) compile hdlc from hq.pm.waw.pl/hdlc 3) uncomment KFLAGS+=DConfig_ZAPATA_NET in zaptel makefile 4) compile zaptel, 5 compile rest

I'm writing a program to quickly generate SIP<Mac>.cnf files for tftp configuration of the 79XX series of Cisco phones and would ask that anyone who is interested in using this please send me working examples of your SIP<Mac/Default>.cnf, and RINGLIST.dat files. Also, please send me your ring tones and * related logos for possible inclusion. Thanks, Christian Hoffmeyer YottaDot

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The

I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 >>>>> > Connected to localhost > Escape character is '^]'. > Asterisk Call Manager/1.0 > Then I type > Action:Login (enter) > Username:sam > Secret:sam > Then I enter twice > > I get Response: error

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

I'm installing TE405P card. This is my zaptel.conf. -------------------------------------- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 loadzone = us defaultzone= us -------------------------------------- When i modprobe wct4xxp, -------------------------------------- PCI: Found IRQ 11 for device 02:0b.0 PCI: Sharing IRQ 11 with

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

Hello. New to the list. We're in the process of deploying Asterisk. Actually, we're going live tomorrow, and I just found out that my Zaptel cards have been mis-configured. I'll preface this by saying that I have looked in the wiki, read through the samples, and attempted to call Digium (they're closed.) So I'm praying that someone on the list can help me out! I have two

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]

We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward calls to PBX and once that is done, I would try to move the telco

I'm ordering a PRI T1 for use with Asterisk and a Digium Wildcard TE405P. The provider is asking me a number of questions about how I want to configure the line. Here goes: 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, E&M, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start -

I once (for a brief period) had dialtone, but I do know why :) Otherwise I get a boooop-booop-booop sequence. I cannot tell if this is the D-Link doing this, or asterisk... Who should be giving solid US dialtone? My indication.conf says: [general] country=us ... [us] description = United States / North America ringcadance = 2000,4000