search for: 1&zap

Hello - I have a TDM400P with 2 FXO and 2 FXS modules. Feeding the FXS modules are two VOIP lines which are terminated by VOIP adapters and have regular RJ11 wires connecting to the FXS ports. Since the two different VOIP lines have different phone numbers, and I know and can tell asterisk which VOIP line is connected to which FXS port, can I cause a distinctive ring on the extensions if a call comes in the 2nd VOIP line? ie, I'd like to ring all...

I can't seem to make the fax detection work. Here's an excerpt from zapata.conf: signalling=fxs_ks group=0 context => guestaccess channel => 47-48 and from extensions.conf: [guestaccess] include => incomingmain [incomingmain] exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24 exten => s,2,Voicemail,u7000 exten => s,3,Hangup exten => fax,1,Dial,Zap/3 When I dial in, Asterisk simply rings Zap channels 1, 9, 10, and 11, whether it's a fax or non-fax call. What am I missing? Thanks, D. --- Martin Pyck...

I just called 6610 from Zap/1 exten => 6610,1,Dial,Zap/1&Zap/2 (Don't try this with the handset pressed up against your ear) ...is this the call waiting indicator? Is it a file, can it be turned down - and does it get used anywhere else? These are both fxs ports. In zapata.conf changinging to callwaiti...

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep 28 13:08:02 NOTICE[20498]: chan_zap.c:5119 ss_thread: Got event 2 (Ring/An wered)... -- Detected ring pattern:...

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

For a *long* time now, I've noticed that the timeout passed to the Dial application doesn't really work as documented. When I Dial(Zap/1&Zap/2&Zap/g3,15,tP) the Dial doesn't timeout for 25 to 30 seconds. I found this problem when I had a 20 or 30 second timeout specified, and it wasn't actually timing out for 40 - 60 seconds. I switched to a 15 second timeout specifically to get a the 20-30 second timeout I wanted....

...urrently assigned to the specific Zap channel, and somehow have it indicate that *all* of the available trunks are in use (all 2 of them right now). This so the people using the system know that both outgoing lines are in use, please wait to make a call. I've seen syntax like exten => 1234,hint,(Zap/1&Zap/2), but I don't understand the & part (does it mean any of?). Is there some similar syntax that means all of? Or, is there a way to synthesize a dummy extension hint out of the individual channel hints? Or.... is what i'm asking not feasible? I hope thi...

...e for() loop would tidy things up nicely. Is AGI what I need to be using? I wasn't sure how to do things such as Dbget(), except through the Exec() call. Here are snippits to show how it was done: /var/lib/asterisk/agi-bin/fln.agi: #!/bin/sh [ $# -gt 0 ] || exit 0; echo -e "Channel: ${1} WaitTime: 1 Callerid: Free Line Notification <(000) 000-0000> Context: default Extension: s Priority: 1" > /var/spool/asterisk/outgoing/fln.$$ /etc/asterisk/extensions.conf: [from-inside] include => to-internal include => app-freeline exten => h,1,Macro(hangup) [check-fl...

...am running last week's CVS build, though earlier builds had the same problem. I originally was using a Pentium II 350, and the problem was much worse, I have since aquired an entirely different system - a pentium II 450, and it is slightly better, but not usable. CPU is about 5% normally and 10-15% during a call, so I don't think this is an issue. People I call do not complain about sound quality, so I guess it is just my side. What can I try next? Thanks Here are some configs: shore-server root # cat /proc/version Linux version 2.4.20-gentoo-r5 (root@shore-server) (gcc versio...

Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house phones. Also, a Wildcard TDM400P, using the wctdm module with 1 FXO module, this supplies FXO to the upstream telco (a single line). The problem: Lately, and witho...

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to...

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment: ==== snippet ==== ; Outbound exten => _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt) exten => _9.,2,Macro(fastbusy) exten => _9.,102,Macro(fastbusy)...