similar to: Call forward on Telco line

Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as "called number", it has to be send as "keypad facility". Anyone here with some hints? The application

I doubt it's possible but I'll ask just in case there's a "legal" way to do that. I have an asterisk server setup at work. When someone call from a PSTN line and enter an extension it rings for a few seconds on the SIP phones of that extension and then if there's no answer it transfer the call to the cellphone of the person having this extension using hook

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

Does anyone have all the Telco intercept prompts (numbers and such) with voice inflections to simulate number referrals and disconnects I could download? TIA, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080522/08b001ce/attachment.htm

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

I am trying to send 'codes' over an isdn2 link - such as *#24# - to activate call forwarding. But it doesn't work. I have tried sending it as a straight dial, and also as a DTMF string...but no luck... I spoke to a telco tech and he said I had to send a facility code....huh? Anyone with any ideas on this one? PaulH

I have an issue with one of my installations running Asterisk 1.4.20 that I need some help with. Not sure if this is new or not but Zaptel or Libpri is not releasing channels properly. I have had issues with calls that stay up for eight hours, long distance on the telco side, so it is more than just a nuisance. Does anyone know if there was a related bug in the 1.4 branch of Asterisk, Zaptel,

I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm

Hi all, Has anyone implemented trunk to trunk forwarding with an asterisk PBX. For the purpose I have in mind its quite important that once the call has been sent onwards to the new desination the lines into the PBX are no longer held. If anyone has UK-specific experience of getting this up and running that would be incredibly useful! Nic

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when

I use a TDM400 card to interface with my telco. I used asterisk voice mail. However, if I'm on the telco line while another call comes in, obviously it cannot go to asterisk voice mail but instead bounces to the Telco voice mail. Is there any means by which I can get asterisk detect the Message Waiting Indicator (MWI) from the telco? Is there some application or variable which can be

Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the

Hi guys, I spent a couple of hours in Google, but the problem appears to be uncommon, so I'd like to ask about it here. The problem is exactly the opposite to "Asterisk does not detect FXO hangup". In my case it's the Telco who does not appear to be detecting Asterisk's hangups. Telco is Telus in Vancouver, Canada. The setup is very simple - Telco -> FXO/TDM400p

I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered.

Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Does anyone have any tips on how to configure asterisk to use the flash and dial codes that my telco provides for transferring calls to outside lines?