similar to: Call waiting question for those who know the source

Can someone please tell me if this is how call waiting should work? ZAP/1 and ZAP/2 are X100P on phone lines ZAP/3,4,5 have phones conected. If a call is in progress between ZAP/2 and ZAP/3 and ZAP/2 gets a callwaiting signal which ZAP/3 hears the Dial command has SIP/2002&ZAP/3&ZAP/4&ZAP/5 If ZAP/4 is off hook, I hear the call waiting signal The CLI says it is ring the 4

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

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

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

Hi, I'm just starting out with Asterisk, in combination with a TDM400, filled with 2 FXS on channels 1 and 2, and 1 FXO on 4. Having just started, all I want right now is to be able to answer incoming calls on a phone connected to channel 1. The trouble is the caller id. I have caller id enabled on my line, my phone supports it, and when I connect the phone directly to the line, it works.

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Help! I am having a problem getting wall waiting to work. I have a X100p and a TDM400 card with and analog phone attached (as well as a number of SIP phones) running on RedHat 9.0 box. The PSTN line has the call waiting feature. If I am on the phone, I get a beep indicating a second call is waiting. However, there is no way, no how, that I can get it to switch to the second line! Flash button,

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

asterisk*CLI> show channels Channel Location State Application(Data) Zap/3-1 (None) Up Bridged Call(Zap/47-1) Zap/47-1 8178062105 at from-nort Up Dial(ZAP/g1/2105||TWK) Zap/25-1 (None) Up Bridged Call(Zap/1-1) Zap/1-1 4999 at from-pri:2 Up Dial(Zap/g2/4999||twk)

Dear List, My problem is that the incoming Caller Id is not displayed on the local analog phones (connected to a TDM400 card). I receive the CID correctly from my telco, but when I place the call to the internal analog line, the CID is not propagated. An interesting point: when I try to place a new call to an already bridged line, I see the second call with the CID on the analog phone. The

I've been working on the local side of asterisk for several days, and I have the in-house dial plan pretty well corn fingered to my satisfaction. Today I began working on the other side to make asterisk do things like place an outgoing call to PSTN and route an incoming call from PSTN. I'm using a TDM11B with a single fxs and a single fxo. My analog handset is plugged into the port

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P <-- analog line --> extension #237 PBX Panasonic Zap/2 = X100P <-- analog line --> extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B

I just called 6610 from Zap/1 exten => 6610,1,Dial,Zap/1&Zap/2 (Don't try this with the handset pressed up against your ear) ...is this the call waiting indicator? Is it a file, can it be turned down - and does it get used anywhere else? These are both fxs ports. In zapata.conf changinging to callwaiting=no stops this. The text implies that this affects fxo ports. John This e-mail