similar to: Q: How do I join an in-progress Zap channel call?

I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!).

I have a couple of questions on the zapbarge: 1) zapbarge asks for a channel - how would a manager know what channel to enter ? Is there any way of being able to enter an extension number instead ? I know that you can get the information from the manager interface, but I wouldn't want to give my users access to this, or have to install / write a system just to get an extension number from a

Here's the scenario: 1. I call out through * using a X100P card to somebody. Then I transfer them to a MeetMe conference and that all works. 2. After the conference is over everybody hangs up but "show channels" shows that the Zap/1-1 channel is still in use by MeetMe and the analog line is not freed up for re-use. Ever. Any clues? Thanks!

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

Hi all, Some simple questions about codecs: What codec does the Zap channel use by default? Can this default be changed, and to what? (g729 too?) What codec does meetme use? (I think this is ulaw, but asking to be sure) Can you use another codec, or does everything have to be transcoded to ulaw? Finally ... if I have a 3way call going, between 1 g729 caller and two other callers, do I need one

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on

How do I monitor a Zap channel as soon as the telephone is off the hook, till it is on the hook again?

I'm having some problems with the volume when bridging two zap channels together. Here is my config..... Asterisk(with TE410) <--> PBX (Toshiba) <--> PSTN (Via T1) When a call comes from the PSTN, in to the pbx, and into Asterisk it sounds great. When a call is sent from Asterisk to the PSTN (via our pbx) it sounds great. When I try to have a call come in from the PSTN, to the

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

Hi, I have an asterisk connected with PRI (Zap channels). If I try to call a number, and recieve cause code 27 because the line 553192 is out of service, but the call continue...is it ok? Here the console messages -- Executing [98 at TRONCAL-PRI-76:5] Dial("Zap/1-1", "Zap/g1/553192") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/553192

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

Hi All, Is it possible to go beyond 250 concurrent ZAP channels with some tweaking or workaround ? Meetme uses zap channels, so we could have a max of 250 conference ports. Is it possible to higher this ? "An Asterisk system can only handle a *max. of 250 concurrent ZAP channels*. This is due to the design limit (255) within the ZAP channel driver." Thanks, ~Vamsi -------------- next

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Sorry about the lengthy post, I've searched high and lo for information on how to do this but now I need your help... ======== Brief intro on problem and requirements =========== I'm hoping to use Asterisk in a Home environment where I'd like to replace the current non-PC Answering Machine, and get added benefits such as IVR, and text to speeach, for Home Automation purposes.

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

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.