similar to: can't create Zap channel

(Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very simple setup. A Linux IBM X4300 Server is running CentOS 5.4 + Asterisk 1.6.2 (one month ago I

Hi all, Does Asterisk support multi thread? I mean: Is it possible to do one of the 2 following scenarios: 1. Play a low background music when the user record his/her voice 2. If the first scenario is not possible, can we play two music stream at the same time? i.e: using MP3Player to play a music file and at the same time play the recorded voice of the user. Thanks in advance for any

My Asterisk fax detection used to work, but no longer does. OK. So, here's the deal: 1. It appears that the "faxdetect" command cannot be applied channel-by-channel in zapata.conf anymore, as Asterisk appears to the last "faxdetect=" command to ALL channels. 2. My stations are detected and sent to the proper extension; i.e., when I send a fax from one zap extension

Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul.

Hi all, I have installed an asterisk server at my office, the server is behind a firewall. On the firewall I've set NAT a rule for incoming traffic on port 5060 to be forwarded to the server. Connecting from home with my sip client doesn't work at all. The asterisk server itself is ok, when I make a local connection at my office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it

I am getting quite frustrated today, so please bear with me. I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk) now my Fedora does not appear to be recognizing my X100P (clone) at all. Hardware browser just shows them as unknown device. driver: hisax So, of course, my zaptel drivers do not work and therefore my asterisk does not work. any help would be greatly

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list

Hi, I'm running Asterisk 1.8.11.1 @office. The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working. No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems Any suggestions or thoughts? Alex Celi This is the info central*CLI> sip show peers Name/username

Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __________________________________

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

Hello All, I have new TDM04B installed and working fine with Asterisk 1.0.5 built on RedHat 9. All is working fine except CallerID that bothers me big time. I have several Panasonic and Sony phones and CallerID works fine with it (when I plug in the line into phone instead into Asterisk I get CallerID) but fails with Asterisk. I am based in California (San Francisco) and my Telco is SBC

Using CVS-HEAD libpri, CVS-HEAD zaptel, CVS-STABLE asterisk. Everything compiled fine. No problems loading chan_zap.so. Incomming calls to PRI work fine. Outbound is a different story: -- Executing Dial("SIP/64.72.107.4-4122fb40", "ZAP/R1d/18005551212|60") in new stack -- Called R1d/18005551212 -- Channel 0/23, span 1 got hangup Mar 13 13:19:29 WARNING[28835]:

I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between

I am trying to analyse recurrent failure times using survfitr from the survrec package. To do this, I need to "Create a survival recurrent object" using Survr. But, when I do this, I get an error "Error in Survr(r1d[, 1], r1d[, 5], r1d[, 6]) : data doesn't match". Here, r1d[,1] is the identifier for each case, r1d[,5] is the time of recurrence, r1d[,6] is the status

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi I have installed 2 fx0 cards into a machine 1 for pstn (out) and the other for pstn (in) However I can get the first card (pstnout to work fine) I have swapped the cards and both work I am also unable to make outgoign on the second card ztcfg -v shows r2d2:/etc/asterisk# ztcfg -v Zaptel Configuration ====================== 2 channels configured. My /etc/zapdata.conf file # fxsks=1,2

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is