similar to: Delay between DTMF Down & Detected Digit

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hi Group, I have an asterisk running as media gateway with a Digium TE405P 2nd Gen rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN Pri. The voice quality is clear except that sometimes a hear a beep sound that occure around 5 to 10 secs in the middle of the conversation. When I check the logs in the asterisk, I found this. Nov 30 00:48:38 DEBUG[27705]

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

Hello, my configuration is the following: a TDM400P board with an fxs and fxo daughter boards on it. I thus connect a fax to my FXS port, after having verified that this port was correctly functioning. For this, I had tried before with a simple phone, and with some basic voicemail exten scripts. Here is my simple dialplan for my fax reception: exten => 300,1,Ringing() exten =>

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Hi everyone, I have a PRI conection on an * system running Asterisk 1.0.9, libpri 1.0.9 and zaptel 1.0.9.2 connected to an AXE 10 (APZ 21220 System 64) in the network side. I know the system and the wildcard I´m using are ok because I´ve used them before with other PRI connections (to a Siemens EWSD) without any problem. First the PRI didn´t work (I got the TE110P alarmed with red). After

Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI with my Asterisk using a Digium TE220B card. The Panasonic is master clock and Asterisk is slave, additionally the Panasonic takes the clock from the PSTN E1. This scheme works and I?m able to make and receive calls the problem is that some times the Asterisk lose the synchronism with the Panasonic and the log shows this

Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a "straight" cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same.

Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'

Asterisk is getting red alarms on my T1, sometimes once or twice a day, but today it happened 5 times. Even once is too many. Every call in progress is dropped. Please help! What do I need to do? What can I try? I've googled and searched this list and can't find anything. Here's an example from the logs: Feb 7 13:37:54 WARNING[32451] chan_zap.c: Detected alarm on channel 6: Red

Hello - I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone =

Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring

Howdy, New installation, trying to connect an RBS T1 with AMI/D4 framing and E&M Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox 2.6.2.1). Outbound calls work fine, but inbound calls fail to read the DID information, and with debug set to 10 I get the following: [Feb 2 19:40:23] DEBUG[25184] chan_zap.c: Monitor doohicky got event Wink/Flash on channel 3 [Feb 2

We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2:

Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The "flaw" in the messages is the "Alarm cleared" message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All

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,