similar to: Catv ATA problem

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi there, I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports. I was wondering if anyone has details about the correct signalling of these FXS ports when connected to original X100p. Tests: fxsks on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing but the zap channel sees no call coming in. fxsls on the zapata.conf and zaptel.conf

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.

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can

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

Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

A client wants to put phones in a semi-public place, using a Calling Card solution. What kind of hardware is suitable? I'm looking for a wall mounted booth, a SIP phone that can't easily be broken, but I might just use cheap analog phones and a channel bank. What do you suggest for the calling card software? This installation will be outside the US. -- Chris Mason NetConcepts (264)

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

Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. I have a TDM31B card. Any help on speeding these connections up would be good - I was on the understanding that if you bridged the channels, then the call should essentially flow straight

I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between

We have a billing system that depends on the CDRs. We had a guest that made a one minute call to a local cellphone, this call went out Zap channel through our channel bank. The CDR recorded a 200 minute call, but I checked with the Telco's records and it had terminated after one minute. What can cause this and what can I do to prevent it? -- Chris Mason NetConcepts (264) 497-5670 Fax:

I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

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

Hello everyone, Our company has an Asterisk server where one of the T1s is connected to an IVR. Asterisk is configured for FXO Loopstart, and the IVR is configured FXS. We can call into it, but our software doesn't expect DNIS digits on FXS, and the digits are being interpreted as user input. We can certainly change our software accordingly, but our test bed also includes other

I have two loop-start phone lines connected to an Adtran TA 750 channel bank. When a call comes into asterisk thru the first phone line and the call is redirect (in transit) back thru the second phone line, the channel bank will leave one or both lines opened (ie hangup with spawn a connection again). does any body have a suggestion: 1) I checked the cables into the channel bank from both phone

HI Asterisk Gurus I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P card as T1 to connect my server to the channel bank (i use a cross cable), the zplex doesn't show any alarm neither the server. I'm trying to make calls from (ext 1) from the channel bank, to other (ext 2) in the same channel bank, but when I dial the ext number, the dialtone doesn't

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

Err, stupid me. Seems like the phone is hungup, but when I start ztmonitor, the line is picked up. I guess the line in India is pretty loud b/c the RX bounces off the charts, changed it to (-4) and all my problems are solved. Regards, Min Chang On 4/14/06, Kyle Sexton <ks@mocker.org> wrote: > > Have you tried putting a Hangup in your extensions.conf? > > > > On 4/13/06,