similar to: Question :: groundstart and loopstart

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel:

hello, I just installed 2 brand new x100p cards into my linux system. Everything compliled and worked fine from the first try. Hoewver the phone lines I will use with this system have groundstart signaling. I have tried to change the zaptel.conf and zapata.conf to use this type of signaling but ztcfg gives me an error every time with the new configuration. Bellow are the config files and the

Hello, I am having problems configuring a TDM400P 4 Port FXO card with groundstart signalling. The box has 2 X100P's, 1 TDM04B and 1 TDM40B. I can configure it for loopstart and it works, just not groundstart, which I need for this installation. What am I missing? Thanks! Mark /etc/zaptel.conf #two X100P FXO Cards # fxsls=1-2 # TDM400P 4 Port FXO (TDM04B) fxsgs=3-6 # TDM400P 4 Port

Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

A thought occurred to me to on how to further quantify the impact of "glare" on a properly dimensioned trunk group and debunk the ground start glare concern. A cursory traffic analysis clarifies: 1. Assume you have a two-way trunk group, dimensioned for average busy hour, average busy season for P01 grade of service, Erlang B or Poison distribution. 2. Further assume that the

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as

Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

I just picked up a couple CAC Access Bank 1s loaded with FXS that should be arriving shortly. Does anyone have one that they use with Asterisk? If so, would you be willing to shoot me a note with your current configs? I'm not very familiar with CAC/etc, and it would save me countless hours of muddling through .. :) Any others tips you can give me on these banks would be very

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Hello! Recently I posted a question about an installation I have that was experiencing glare problems. The solutions presented were to use inverse inbound and outbound line groups and to use groundstart signalling. As it turns out, the Sangoma A400D card that is in use does NOT support groundstart. I've confirmed this with a Sangoma engineer and their support staff. I've also read that

*** Now you just need a "phone" or "station" (FXS) device and a method of connecting it. *** So... essentially I need to spend another $125 US on a Wildcard TDM400P so I can plug in an analog phone? On the X100P description it says: "The X100P supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). It can detect ringing and

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

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