similar to: Voicemail Hangup detection issue

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Hello. I am running asterisk 1.2.14 on a Dell poweredge with a Digium FXO/FXS card connected to 6 analog lines and using Linksys spa942 phones. My users are complaining of randomly disconnected calls, and when I watch the log (debug warning,notice,error), I don't see any cause. It looks like asterisk is seeing a hangup from the analog end. I have attached my zaptel.conf and zapata.conf.

> niles@atheos.net wrote: > > Christopher, > > > > Just as a total guess, check to be sure the PoE portion > >of his cat5 cable are not either grounded, or touching > > each other. Also, be sure he's not connected to a PoE > >capeable switch since cisco is reverse polarity of the > > PoE standard. > > Niles Niles, Thanks for the guess. We

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,

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

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

Hello, I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel => 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling requested on channel 4 is FXO Loopstart but line is in FXS

Hi All, I've got an MD3200 modem which is working as a Zaptel FXO interface for Asterisk (X100P clone I believe). It seems to work, but on incoming or outgoing calls I can hear the other party ok but when I speak, I hear my voice echo back at me (quite quietly but it's distracting!) on everything I say. The other party doesn't seem to hear the echo - they just say I sound a bit quiet.

Hi, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. ------ Salvatore. ----- Original Message ----- From: "Sasa" <sasa at shoponweb.it> To: <asterisk-users at lists.digium.com> Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card > Hi, I would like remove a Digium TDM2400P from

I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register

I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

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.

Okay, I'm going to preface this by saying I'm sure I've overlooked something really basic here. I just need someone to hit me with a clue stick and point out what I'm missing. I've got a TDM card with four FXO modules. I've plugged one of them into a PSTN line. I'm working through the examples in the Asterisk Documentation Project guide, but I can't get Asterisk

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

Hello All, I have this strange problem, I can dial out with my sip phone and it seems to work relatively well, but when I call in, the line just rings and rings, I get no indication in asterisk that it's detecting an incoming call. The strange thing is that in ztmonitor 1 -vv the rx volume goes from around 99% when the line is idle to 0% as soon as the line starts ringing. I'm located

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

I have a system setup with two T1 circuits, the customer complains of call quality issues. I've checked logs and don't see anything too strange, but when I run "dahdi show status" I see: Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1 OK 0 0 19 ESF B8ZS YEL 0 db

My current config: pstn -> audiocodes fxo gateway -> asterisk -> xlite every fxo ports are registered with asterisk I have this extensions.conf exten => 111,1,answer exten => 111,n,dial(sip/fxo1) exten => 111,n,hangup If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a phone no and connect to the called party. this is a two stage

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [