similar to: Incoming voicemail and dialtone

I have some strange behavior happening with call flow when analog line errors are encountered. This may be due to the way that the X100P detects "busy" signals, or it may be something in the software. Could someone with more in-depth knowledge make a comment on the items below? My dialing logic says "dial local area code numbers out of the analog line, and if the analog line

> -----Original Message----- > From: Asterisk@gtcus.com [mailto:Asterisk@gtcus.com] > Sent: Saturday, September 06, 2003 8:39 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone > > > Thanks for the great feedback on these options. I am fairly > new at this and not familiar with the IAX/IAX2 capabilities > offered by

Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then

Hi, I wish to know if GNUGk can work with * running as a gateway with the Digium FXO card. Kindly share your experiences in case there are some issues which one must know before going in for such a setup. Also, I've been reading about the DialTone detection capability by the hardware in different countries. What are the issues with it? Thanks & Regards, Deepak ----- Original Message

Does anyone know if there is a zapata.conf option to tell * to listen for a dialtone before dialing? I've got a couple of analog phones on a pstn line shared with a x100p * fx line. If someone is on the analog phone and another person initiates a call through * to use the same line, * dials over the top of the existing conversation. Is there a way to have * detect dialtone before dialing?

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Is there any way to make the X100P to check for a dialtone before dialing? My roommate is tired of hearing random dialing in his ear when he's on the phone (same circuit as the X100P). I know one solution is to just put in a TDM400P for the house, but I'd rather get away w/o having to do that. Thanks, Jared

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

The Vonage service is offered with a SIP Cisco ATA device for connection to an analog phone. Is it possible to connect the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO card? Are there other services that offer this capability or something similar to IP dialtone? Thanks Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi A quicky - has anyone had success with an X100P in the UK ? I'm seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016

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

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hi there, i have a Problem with dialtone and a TE401P Card. I swear I surfed the wiki, the mailing list and google for 4 hours and did not find the solution, can you help me ? In Germany I have an E1-Line and an Alcatel 4200 PRO PBX. Without using asterisk I dial the "0" on an Alcatel Phone and have the local exchange dialtone, then I can dial. Most users do not dial en block, they

I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? </edg>

Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the

Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

Is there an easy way to get information about local calling areas out of telcos? I'm trying to get a list of area codes and prefixes in my local calling area out of Verizon, and it looks like they've stopped providing the information online. Is there an easy source that I'm missing, or do I need to call them and have them mail me a copy every few months? Scott

I am having a problem with your my nufone service. I'm trying to setup incoming calls and I'm having no success. Outgoing works fine though. The message I'm getting is "the person you are call is not currently reachable". I'm going to give you as much info as I can. I'm also an asterisk newb! Anyways, I installed asterisk@home. Set up extensions which communicate