similar to: asterisk 1.4 and zap channel flash

I am using asterisk CVS 10842 and a TDM 400p withanfxs and an fxo module and when I dial the fxs channel it rings for a second and then says no answer after 20 seconds. I also have the latest Zaptel drivers. Here is a log snippet. Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Executing Dial("SIP/15712523171-09c1", "ZAP/1|20|trwW") in new stack Feb 25 00:53:11 VERBOSE[2015]

Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25

Hi. I am using 1.2.5 and I have an extension using a zap fxs channel on a 400P Digium card. Now when thatextension is dialed with a timeout of 20 seconds it rings for about half a second and then the log says noone picked on after 20000 seconds and so it goes to voicemail. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do

Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box

OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an asterisk from the other end of a call, I here it forever till the call hangs up. Looks like it starts the vldtmf, but never ends it from the logs. I am using Digium 400P rev I with one fxs and one fxo module. Any way around this one? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How

Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi, i'm new to * I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; when i try to call outside i get: -- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2[pippo@pippo]/2", "Zap/g1/0123456") in new stack Jul 13 13:42:49

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1

Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do

A bit about my experience with the TDM-04 FXO. Only saw a few post on this subject, thought I would contribute a little about my experience to save others the hassle. a. As an earlier poster noted, the driver for the FXO is in the wcfxs module. Perhaps it should be renamed to something less confusing. b. You need the zaptel,zapata libraries from the cvs, the ones with Asterisk 0.7.2 won't

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

(Sorry of this appears in the list twice, but I wasn't sure if it was blocked or not) Hi there, I'm having a problem with my TDM-400P which has been working like a charm up until very recently. It started to fail last week, and so I was hoping someone could illuminate me with some information as to why. Its configuration is as follows: ------------ FXS (green) module is in position 1,

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Hi. Ever since I bought my Digium 400P with 1 FXS and 1FXO module, once in a while I hear what sounds like a touchtone in my ear on a phone hooked up to the FXS module. This was not heard by the other side, and although it was annoying, it was not too much of a problem till I was using the asterisk 1.4 (rev 46317) and the beta of zaptel 1.4 (rev 1536). Doing this, the toutchtone noises once

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

As everybody must be watching the superbowl. I post this to let you have some fun while thinking what this can be. TDM400p (fxo) connected via loopstart to ports in an AvayaG3 call comes in from the avaya to the tdm card: WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' but call can be processed normally. comments? --

Hi. I was using asterisk 1.2 on a box with sip phones attached and a long distance T1 line as the phone provider. We did a successful test of *1 allowing one-touch recording as set in the features.conf. Because of deadlock issues I decided to try 1.4 (latest svn as of yesterday) and the deadlock went away, but when we tried to use the *1 it was sent over the bridged channel rather than being

Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board