similar to: Help with dialing out with TDM400P

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Hi, I have postgresql and * all up and running as the latest cvs-250205, although something weird. Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit

Calling out on PSTN over a TDM400P seems to generate a Red Alarm - whatever that is. I have another extension on the PSTN, and I can dial out over that. zttool shows no alarms. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call:

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and

R-masters, I have a problem that I have been working on for a while and it seems that there may be a simple solution that I have yet to figure out, so I thought that I would venture to post to the help list. Let's say there was a data.frame with three vectors, two that are factors identifying the data, and one that holds the frequency of occurrence (the events are binary, yes or no). I

Hi all, I am new to this group, I am using Prototype''s Ajax.Request o make ajax call but i was not able to invoke my backing bean method. any one help me please. My jsf page: <%@ taglib uri="http://java.sun.com/jsf/html" prefix="h"%> <%@ taglib uri="http://java.sun.com/jsf/core" prefix="f"%> <!DOCTYPE HTML PUBLIC "-//W3C//DTD

On Thu, Sep 14, 2017 at 3:08 PM, Stephen John Smoogen <smooge at gmail.com> wrote: > On 14 September 2017 at 14:51, Larry Martell <larry.martell at gmail.com> wrote: >> On Thu, Sep 14, 2017 at 2:32 PM, Ulf Volmer <u.volmer at u-v.de> wrote: >>> On 14.09.2017 19:54, Larry Martell wrote: >>> >>>> Where would I do that? This is something

I was trying to authenticate from AD and could not bind with the server. I was able to do ldapsearch. please suggest the possible cause for failed to bind server. This is the log info from log.winbind-idmap dd_dual.c:60(child_read_request) child_read_request: read_data failed: NT_STATUS_END_OF_FILE [2018/04/19 20:03:09.741686, 0] ../source3/lib/smbldap.c:1046(smbldap_connect_system)