similar to: zt hook failed

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hello, we are setting up another system that will run either 1.2.4, the latest version of 1.2 or 1.4. We have not yet decided on the version. Anyhow, this is a higher volume system (dual processor) which will handle 30-50 simultaneous calls with 60 to 100 simultaneous channels lit up. Most calls are g711 with very little g729 and a little gsm mixed in. We have a similar system doing exactly

I have a small system based on one TDM400 card with - 2 FXO modules for incoming lines - 1 FXS module for one phone The system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS

Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and zaptel (1.2.5). The problem is that 6 modules can't be initaized and i get the folowing error: May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No such device May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such device here = 0, tmp->channel = 9, channel = 9 ztcfg output

I recently purchased a Asterisk Developer's Kit (TDM) and now have it outfitted with 2 FXO modules and 2 FXS modules. I'm not using the X100P modem card that came with the kit. I'm having problems with dialing out on my POTS line. Successful dial out is intermittent. About 50% of the time the call goes through. The other 50% it is dialing the wrong number. ( I can hear the error

Hey guys, We have asterisk 1.2.7.1 running and today i got following error message. and users started complain regarding call issue. after reboot everything comes back to normal. I just want to know what happened ? Feb 25 11:27:11 WARNING[10042] app_meetme.c: Error setting conference Feb 25 11:27:11 WARNING[10042] chan_zap.c: Unable to set audio mode on channel 1 to 1 Feb 25 11:27:11

Greetings all, I'm new to asterisk... Finally got two IP phones and two TDM400P's (one with four FXO modules and one with four FXS modules)... Began experimenting a couple days ago and have ran into a very aggrevating problem. My system consists of the following hardware: Abit BL7-RAID motherboard Pentium 4 1.6GHz processor 512MB SDRAM 2 TDM400P cards 2 3Com 3c905

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter

I'm having various issues that may or may not be related to each other (I'm pretty sure they are). We've had this system for a year now (quad T1 card, right now we have 1 T1 coming in, 2 going out to channel banks) and we've had intermittent "ghost" calls--it appears that what is happening is a call is made, the number being called is disconnected/busy/answering

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

Every once in a while one of my FXO interfaces will not accept incoming calls. The other interfaces on the same board will continue to accept calls. Here is what I see in the logs: ERROR[4465]: fsk_serie made mylen < 0 (-22) WARNING[4465]: CallerID feed failed: Success WARNING[4465]: CallerID returned with error on channel 'Zap/3-1' I also see this: WARNING[811]: zt hook failed:

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

Dear group, Was wondering if anyone out there has had the experience I have been having. In reading recent posts on echo cancellation, I think there is.... We recently cut over the Asterisk and are configured with 5 FXS and 2 FXO ports to the PSTN via 2 TDM400P's and 5 SIP phones on our local network. I have set up echo cancellation with 800ms echo training. I do not have

Hello, I'm trying to develop a box plot of time series data to look at the range in the data values over the entire period of record. My data initially starts out as a list of hourly data, and then I've been using this code to make this data into the final ts array. # Read in the station list stn.list <- read.csv("/home/kbennett/fews/stnlist3", as.is=T, header=F) # Read in