similar to: zaptel hardware detection with genzaptelconf

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

Hi everyone. We have a problem here... Hope somebody can give us some hints. We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem. Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and libpri (1.4.3) is installed. There is a QuadBRI-Card installed: # lspci -vv -s 06:04.0 06:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01)

Hi Version 0.9.0 of Xorcom Rapid Debian/GNU/Linux/Asterisk has just been released. Main changes: * A decent version of Asterisk/Zaptel (1.0.2) is provided * Includes a better default configuration * Automatic detection of the most common Zaptel cards * Contains more optional software (apache, MySQL, Webmin and others) - available via menu You can get it from:

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'm having some trouble setting up a Xorcom Bri 4 port. I have compiled asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches. So I'm running zaptel-1.2.17.1 and asterisk-1.2.18. The problem I'm having is that for one I get no LEDs showing if the unit is in TE and NT mode (not a issue for me but may have some impact on things) I have no errors in any logs I

Hello, I have an Asterisk box with a digium TDE205p card. The problem is that I have several "Goto(s-${DIALSTATUS}" sentences and while the call is trying to find a free channel to establish the call, I get a little noise in each "Zap/... is proceeding passing it to Zap/..." line. My configuration is: Digium card: TE205P Asterisk Version: 1.4.21.2

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

On Sat, Jan 17, 2015 at 09:31:33AM +0100, Bertrand LUPART - Linkeo.com wrote: > >>> However, dahdi_genconf keeps finding no span: > >>> What am i missing? > >> > >> It looks like your driver is loaded correctly. My guess would be maybe > >> the dahdi-tools is packaged as an older version that doesn't know > >> about the newer te435

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Can anyone describe the asterisk implementation of this any better than the sample config files do? from zapata.conf ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group => <trunkgroup>,<dchannel>[,<backup1>...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running

Hi, I use asterisk branch 1.4 and gui 1.4 as well. I have the following situation: When I try to make gui configuration by http://localhost:8088/asterisk/static/config/setup/install.html I can see that application logs my user correctly but there is no browser window shift to the next page. it stays at the logging one. I get the following info in the console: [Feb 4 09:33:09] == Parsing

hi every body; i installed zaptel 1.4,libpri 1.4, asterisk 1.4, asterisk-addons 1.4 succefuly, but i can't find the command "genzaptelconf, so i tink to install "handy zaptel toolset" please can someone tell me whitch package goes with zaptel 1.4, i consult "http://updates.xorcom.com/rapid/pool/main/z/zaptel/" but ????????????

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

Hi folks Xorcom Rapid 1.0 is avilable for download Q: 1.0? A: Sure, better than 0.9.1: * Asterisk 1.0.5 * Base packages upgraded * Built with SpanDSP support * Improved Zaptel detection * ast-cmd with some useful command-line abilities provided * ssh installed by default * putty.exe is included on the CD * music-on-hold files removed due to potential licensing issues

About the IVR, are u using Asterisk? Regards Bilal > ------------------------------ > > Message: 17 > Date: Wed, 18 Feb 2009 12:23:41 +0200 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] Credit Card processing > machines > To: asterisk-users at lists.digium.com > Message-ID: <20090218102341.GD21440 at xorcom.com> >

Hi I put a snapshot of our current packages updates.xorcom.com . They are available from the deb source deb http://updates.xorcom.com/test sarge main (this is s/rapid/test/ of the name of the source of the stable version) Changes include: * Fixed and simplified zaptel detection * Support for spandsp: compiled but not yet tested * IAX extensions * Mail server configuration * The script