similar to: facilityenable in zapata.conf

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

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

Hi, we are using asterisk with Junghanns QuadBri and some sip phones. 2 channels are configured in NT mode (ISDN PBX connected, internal ) and 2 channels are connected to the public ISDN network (bri-cpe). We use Bristuff 0.2.0 RC8C from Junghanns. When a call comes in from the public phone for a specific extension (Hotline Number), we initiate a parallelcall to some SIP phones and also to our

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

has anyone an idea how to display incoming national/international isdn-pstn-calls correctly to internal isdn AND sccp/sip-phones ? without nationalprefix=0 and internationalprefix=00 I get incoming phone numbers correctly on isdn-phones but the leading zero's are stripped of for non-isdn phones when I set this prefixes inside zapata.conf my internal isdn-phones get this prefix twice...

Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a "0" as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success:

I have a issue with two Zap clone cards where they used to work. I am using Asterisk@Home 2.5 which includes Asterisk 1.24 and Zaptel drivers 1.2.4. The system is a new Intel Celerion machine. I used to have the same cards running in a Intel PIII system. in this system, they worked. In this older system, I was able to call into the machine and call out from it. Now that I have upgraded

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

I have a Digium TDM400b11, 1FXO [port2] & 1FXS [port 1] When I reload the chan_zap I get: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be specified before any channels are. Apr 9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: load_module

You should go here and enter your suggestion so the developers can track it. http://sourceforge.net/tracker/?group_id=123387 Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aaron Glenn Sent: Thursday, March 10, 2005 4:38 PM To: Asterisk@Home Cc: Asterisk Users Mailing List - Non-Commercial

From: Malcom Kemp Sent: Wednesday, April 25, 2007 11:19 AM To: 'asterisk-users@lists.digium.com' Subject: AsteriskNOW generation of zapata.conf file I am a new user of Asterisk, and an trying to use AsteriskNOW. The test system is dual processor, so I am using the beta 4 version. I am currently trying to manually configure, as the GUI does not seem to let me accomplish what I need.

Another dumb question... My asterisk system seems to have problems detecting hangups. I am getting a LOT of voicemails with dialtone or silence. I see over at asteriskguru.com there is an explanation of how to configure for polarity reversal in zapata.conf? Does zapata.conf have any function in systems that aren't using zaptel( I suppose not)? I am using an external gateway (wellgate

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO modules) and a TE205P dual E1/T1 card. The problem is with the TE205P and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf files. I go in and add the required lines to configure the TE205P and restart Asterisk. All is fine until I reboot the

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

Hi pals, Pardon me if this question sound basic please. This is the first installation where I have to use the analogue card and therefore a little lusty. I have googled a lot, but though there is a lot of information about the above file, none indicate where the file lives. I have a installed asterisk and zaptel software on a fresh installation of CentOS 5. This all from source and following

Hi asterisk GURUs I have an issue with a call forwarding setup, I have the next zapata conf: -----------------zapata.conf----------------- [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 channel=1 -----------------zapata.conf----------------- And in extensions.cof I have a

Hello I couldn't find information about what "wink" is in zapata.conf: www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters Does someone know what it is, and how it differs from flash? Thank you.