similar to: feature transfer on PRI

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

I am having a problem with SpanDSP. What happens is when I send a fax to SpanDSP the fax message seems to fail in the training phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

I am having a problem with SpanDSP. What happens is when I send a fax to SpanDSP the fax message seems to fail in the training phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind

When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk

Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a

Hi list, Just installed spandsp. In my limiting testing, I have an issue on a Philips fax machine (HFC21) directly connected to my * server through TDM400, reception with rxfax works fine, but txfax always fails. Below is a transcript of failed transmit. This is with asterisk-1.0.3 (with native moh patch but I don't think it is the source of the problem). I already tried libtiff 3.5.7,

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == ==

Today is the first time I've had any show stopping errors with asterisk since I started using it over 8 months ago... While I wait on the telco to check the circuit I thought I would post this here. Not sure yet but I thinks my 4-port PRI card is gone!!! I made a call and right when the person answered - BOOM. This is only a little better than it not working at all.

Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We