similar to: Delay on zap channel

Hi, I've a wildcard TDM400P card with 2 fxo and 2 fxs modules. I've set this extension in my extensions.conf for obtain the external line: exten => 0,1,Dial(Zap/g2,10) The dial application is executed immediatly but next this there is a delay before I can hear the tone. This is the output in the CLI: -- Starting simple switch on 'Zap/1-1' -- Executing

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

Sorry for the very newbie-like question. I have the FXS part straight. The part I don't understand is the FX0 part. Will I need the FX0 card if I am connecting to a service like FWD? My goal is to get rid of my phone line all together. I am under the impression I will only need an FX0 if I'm connecting to the central office side of the phone connection or to an existing PBX. Bottom

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hi, After using Asterisk with a SIP hardphone for a couple of weeks I've just installed a TDM400P card. My hardphone - a 7940 - allows me to use a dialplan to decide when a particular extension is complete and automatically trigger dialing. This works well with my internal extensions, which are all of the form "Z00". When trying to dial these extensions from a handset

Hi there Background: I want to add DDI and voicemail to users on an existing analogue pabx.. It does not support ISDN. I have 10 DDI numbers via IAX which I am having sent to my Asterisk box. I have 2 X100P cards connected to 2 analogue extension ports of my main legacy analogue pabx. I have set up voicemail for each of my DDI numbers, and when a call comes in for the person at pabx

Hi, I am trying to do the world's most simple install. I have a Wildcard TDM400P with 3 ports: 1 FXS on port 1 and 2 FXOs on ports 3 and 4. (i'm not using port 3 for now, put want it for expansion purposes) I simply (to start with) am looking to have the FXS phone ring when an FX0 port is dialed. I would also like to be able to place outgoing calls on the FXS through the FXO. Right

Dear group, Was wondering if anyone out there has had the experience I have been having. In reading recent posts on echo cancellation, I think there is.... We recently cut over the Asterisk and are configured with 5 FXS and 2 FXO ports to the PSTN via 2 TDM400P's and 5 SIP phones on our local network. I have set up echo cancellation with 800ms echo training. I do not have

Open the console with verbose turned up. Make a test call and see where it is hanging. That will isolate the problem. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo > Sent: Thursday, April 27, 2006 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Is it possible to detect SIT tones on an outbound call? Specifically this is for an outbound call generated via a .call file over a PRI. I get answer supervision when the SIT tone starts and Asterisk believes this is a successful call. I'm using Asterisk 1.4.6. Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes with a Philips fax machine. It seems that the fax machine doesn't recognize the carrier. How can I see the spandsp logs? I've enabled debug on the asterisk CLI, but I can't see any output while the txfax/rxfax application runs. Stefano Arata.

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this

I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS). These connect to a Digium TE210P card. I'm running kernel 2.6.10 and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today. The results are the same for all versions: Right after I reboot, and modprobe wct4xxp, my analog phone connected to port 13 of the first channel bank (first FXS port) gets a dial

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a