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Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

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 list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? Thanks, Jaap PS -- Below is some info regarding my configuration. =========================== Zaptel version: 1.4.7 (incl. firmware and modules). OS: Debian

Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down

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:

Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down

Asterisk Version: 1.2.9.1 Zaptel Version: 1.2.6 LibPri Version: 1.2.3 Hey List, we are running an asterisk server in connection with an octopus telephone system. I have expired some random drops of zap channels bridged to SIP Telefones ( snom 190 ). Asterisk Messages shows something like that: Jun 9 09:32:33 WARNING[3207] res_features.c: Bridge failed on channels SIP/fon01-6945 and Zap/32-1

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

Dear All, I am facing a strange problem that I can't find any matches for while googling, sometimes while a call initiated from asterisk to the PSTN is answered and the answering party say the receiptionist tries to transfer the call to someone else, the call dies, the full log shows nothing useful and I am really unable to move forward on this issue, so can some one suggest anything? My

Hi Francois, I kwnow, but I have "DSP:on" also if i not have an hardware echocan module :/ and I always have "Echo Cancellation: 0 taps, currently OFF". This is my zapata.conf [channels] language = it usecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes cancallforward = yes callreturn = yes switchtype = euroisdn

hi. could somebody explain how exactly the following parameters in zapata.conf work: pridialplan prilocaldialplan internationalprefix nationalprefix localprefix privateprefix unknownprefix the wiki & comments doesn't quite explain them. and phone companies are absolutely no help. i've setup systems in the US & China with trial & error until it works. now i'm setting up a

Does anyone know, how I can increase the timout value for the ISDN implses, so asterisk waits for the extension on the ISDN channel? Regards, martin My zapata.conf: [trunkgroups] [channels] language=de pridialplan=local prilocaldialplan=local nationalprefix = 00 internationalprefix = 000 ; trust user provided callerid (clip no screening)? pritrustusercid = yes ; hidecallerid=no

Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes

My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message "the person you are calling, is currently unavailable" This is an ubuntu machine, with a sangoma card, with

hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

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

Hello, I'm having problems making international calls from our asterisk using an ISDN30 in the netherlands. Below is the zapata.conf that works for all national calls, but international calls all fail a RC=41 ; zapata.conf [trunkgroups] [channels] language=nl signalling=pri_cpe switchtype=euroisdn callerid=asreceived ;pridialplan=unknown ;prilocaldialplan=unknown immediate=no

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley