similar to: Kewlstart - Busy signal before battery drop.

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave

Hey All, AT&T is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,

I am having trouble getting my T100P and ADIT 600 talking. I have 2 FXO 8 port cards in the ADIT and a T1 controller card. I have a T1 x-over cable connecting two together. Here are copies of the relevant bits of my zaptel.conf and Zapata.conf Zaptel.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us Zapata.cfg context=default signalling=fxs_ks group=1 channel => 1-24 My

Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support

Hi All, Which is the best signalling to use when connecting an FXS inteface on a TDM400 to a standard telephone. I see that all examples use fxo_ks, but it is my understanding that kewl start is really designed for connections to the CO so that hangup etc. can be detected. So does it make any sense to configure a telephone for fxo_ks? Or should it be configured for fxo_ls? Regards Garry Taylor

Hello all, I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and I have a question about timing parameters. By my understanding one machine should be the source of the timing and the other a slave of that timing. So on machine A I have the following in system.conf: dynamic=eth,eth0/00:0C:29:55:89:7E,24,0 On machine B I have this is system.conf:

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

>> I'd have to agree with the channel bank plan. You don't even need to >> buy a used channel bank. You can get a new Rhino for $1500: >> >> http://www.channelbanks.com/ > > I gather you've had experience with Rhino. > > How does it work with Asterisk? > Does it provide all features? Caller-id, echo cancellation, how's the > voice

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

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'm configuring a box with a TDM400P with 2 FXS and 2 FXO. I configured the FXO's first to try them, and they worked (I could talk to myself thru the PSTN. But when I add the FXS's to zapata.conf and restart *, I have a problem... [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 4, FXS Kewlstart

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

I hear momentary audio drop-outs (silences) in some flacs I have encoded. These occur at the same time as "wasted bits" messages in my mplayer output. [flac @ 0x860fb00]coding type: constant [flac @ 0x860fb00]1 wasted bits [flac @ 0x860fb00]coding type: constant [flac @ 0x860fb00]6 wasted bits Is there a way to prevent these at encode-time? If this is not the appropriate list for

Hi, I recently purchased two X100Ps and a TMD40B. I have set everything up according to the instructions however I can't get asterisk to pick up an incoming call (It just keeps ringing). Also, when I pick up an internal extension I get dead air. I searched the mailing list for 3+ hours -- What am I missing? Thanks in advance for your help. Here are the installion steps that I used: