similar to: Unable to create Zap channels/IAX Warning

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hello list, I've posted my problem on BSD list and i still have the problem. The remote side receives the call , but there's no voice on the call. I tried everything about possible NAT problems .. but ther're on same net. My platform: FreeBSD 5.2.1-Release Asterisk 1.0-RC2 soft phones : X-Lite >>>> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20")

Hi Group, I have a problem with my FXO interface. It seems that asterisk cannot see any configured channels, only pseudo. Pls see the error and my config below. Rgrds Markus ----- May 2 15:09:00 NOTICE[17327]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Please can someone help me to install chan_capi on Mandrake 10. I get page after page of errors and can not seem to find detailed install instructions anywhere.

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:

I'm trying to produce some enhancements to one of the applications, and am trying to use ast_request("zap", format, "pseudo") to create a new channel on /dev/zap/pseudo, which I can then bind to a zaptel conference and play a stream to it. I've been using as inspiration the Radio Repeater app, app_rpt.c, which uses this technique to play idents and announcements.

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

What I am trying to do is the following: A call is sent to the * box via a SIP invite. The * box answers via an IVR menu system with " Enter the extension you want to dial" so I enter in my 5 digit extension and get the below message. Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No channel type registered for 'SIP)' Jan 18 10:10:03 NOTICE[-1380238416]:

Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing

when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) any explanation for this? Thanks, -------------- next part -------------- An

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

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

I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found ==

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

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse