similar to: TDM, loopstart and modules GSM Nokia32

Hello, I downloaded asterisk 1.0.5, and the cvs version, and none of them has the attended transfer option (x and X). I see a patch here: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 but this patch doesn't applies well in cvs version or 1.0.5. How can I make attended transfers work? It's a feature really needed :( Thanks in advance :) -- Paco Brufal

Hello, I have an Asterisk system with kernel 2.6.18.1 and one quadbri. I have installed the latest bristuff patches (0.3.0-PRE-1s). The system works fine, but when I do a "reboot", the system hangs unloading module "qozap". Is there any known problem with latest 2.6 kernels and qozap module? Thanks. -- Paco Brufal <pbrufal@servitux.es> ServiTux Servicios

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you

I am connecting a t100p to a b8zs, superframe, loopstart t1. Previously I've attached to e&m wink and pri lines with no problems; however I seem to be missing something. Should it be fxols in the zaptel.conf (smartjack to x100p) or fxsls? With e&m wink the dnis set was 100, so it was easy to make an extension 100; and it would answer incoming calls. How do you get a loopstart to

Hello Astrites, I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines. I had tried many configurations. I had attached a toner to the line to see if I was getting this "disconnect supervision" signal after hang-up. (toner has a line powered light in the off position) It seemed that I was getting the signal,

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

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: >---

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 find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi guys, I have a strange problem. I can authenticate a user with wbinfo from my domain controller (security =3D ads), however when I try and map a share, the authentication fails. i.e. # wbinfo -a 'COAL+bcanglo%bcpass' plaintext password authentication succeeded challenge/response password authentication succeeded # smbclient '\\xxxxxxx\timtest' -U 'COAL\bcanglo'

Hi guys, I have a strange problem. I can authenticate a user with wbinfo from my domain controller (security = ads), however when I try and map a share, the authentication fails. i.e. # wbinfo -a 'COAL+bcanglo%bcpass' plaintext password authentication succeeded challenge/response password authentication succeeded # smbclient '\\xxxxxxx\timtest' -U 'COAL\bcanglo'

Hello I have installed my new E100P card in the same machin with two TDM04B cards. I have followthe instructions on http://www.digium.com/index.php?menu=configuration#T_E100P_PRI This is my zapata.conf span=1,0,0,cas,ami fxsls=1-15,17-31 dchan=16 fxsks=32-39 loadzone = us defaultzone=us I do *modprobe zaptel wct1xxp wcfxs ans seems to work ok. * But when i do ztcfg -vvv i got Zaptel

My laboratory is measuring the abundance of various proteins in the blood from either healthy individuals or from individuals with various diseases. I would like to determine which proteins, if any, have significantly different abundances between the healthy and diseased individuals. Currently, one of my colleagues is performing an ANOVA on each protein with MS Excel. I would like to analyze

*** Now you just need a "phone" or "station" (FXS) device and a method of connecting it. *** So... essentially I need to spend another $125 US on a Wildcard TDM400P so I can plug in an analog phone? On the X100P description it says: "The X100P supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). It can detect ringing and

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg

Hi, My asterisk setup was working when using a Wildcard T100P. I recieved incoming calls and was able to make outgoing calls. I recently decided to change my card to a TDM01B one (one FXO). However when I dial a working extension, I can see on the console that the call goes through sucessfully, but it is never actually executed (i.e if I call my mobile phone, Asterisk shows the call as being

Hello all, recently i stumbled upon the I-TDM standard, e.g. see here http://www.picmg.org/v2internal/news2005.htm "SFP.1, also known as I-TDM (Internal TDM), is a companion protocol specification to SFP.0 that is optimized for TDM traffic over high-speed fabrics such as 1 and 10 Gigabit Ethernet (PICMG 3.1), Advanced Switching (PICMG 3.4), Infiniband (PICMG 3.2), etc. SFP.1 and SFP.0

Dear All, I'm using a TDM 2400 (12 FXO / 8 FXS) and I've some issues randomly : When receiving a call from PSTN and after Hang-up, Asterisk detect a second arrival call (same that channel) that is a ghost call off course. I don't PSTN suppervision with my telco... Any suggestions ? Flo_turc

Hi On an old IBM Netvista thinclient, the TDM card doesn't detect incoming calls, although the card seems to be detected, and correctly configured: pbx asterisk # lspci 00:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel