similar to: pseudo Direct Outward Dial

I have two call waiting problems. I have a POTS line into and FXO port and telephones on an FXS port 1) I can't seem to use the flash button(on the phone) to answer a call waiting call. I see the callerid coming though and here the call waiting tone, but I just can't seem to answer it. The flash button seems to have no effect. I have: callwaiting=yes in zapata.conf

I am having dial out issues and was hoping someone could shed some light. The problem is Intermittent: extensions.conf [globals] ; Trunk Info for outbound calls via PSTN - See the zapata.conf file in /etc/asterisk TRUNK=ZAP/G1 ;Trunk Interface ;MSD digits to strip (usually 1 or 0), 1 = remove a leading 9 TRUNKMSD=1 ; -------------------------------------------------- ; [trunklocal] - Defines

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hi all, I am either going daft or not reading things right. I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter. I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards. But, if I pick up the handset to make a call all I get is the engaged tone and the following message.

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

Hello all, This is my first post to this mailing list. I've been following the development (not as a developer, but as an interested user) of Asterisk for over a year now. It's only in the last month that we've decided to give it a try (hey, might as well...everything else we use is open source). We ordered a TDM11B (1 FXS, and 1 FXO). The card and Asterisk install was uneventful.

How do I set asterisk not to answer incoming PSTN POTS calls? I want to be able to use the line for outgoing calls only. -Thanks Tim

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hi All, I've had a great time exploring * on & off over the past month and have had a deal of success, very impressed with this product... I have an issue on a debian unstable system with the zap interface. I've compiled and loaded the ztdummy successfully, but when attempting to connect to a conference room I get failure. I get the error chan_zap.c:757 zt_open: Unable to open

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

Hello all, Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the "s" extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? With a

The subject says it all. Is it possible, code wise, configuration wise, at all - to reverse the order in which the available zap channels are used for *outgoing* calls? Code wise, I looked at the channel structure and it appears as though there is only a next pointer, not a previous pointer, so to 'easily' to this in the code would require a change to the code that reads in zapata.conf?

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had