similar to: I can't resume a call on hold from zap device

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

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

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

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 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

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

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

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

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

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

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

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

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

back to another ? Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0

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:

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hi, I am using asterisk 1.2.5 and have some problems with asterisk connected with an E1 card to our PRI. Dialling in and out generally works. When someone dials in from a mobile phone, all numbers are sent as a block, and the called extension rings as intended. when someone picks up his phone handset, waits for a dial tone, and then dials in manually, the call will be redirected to the

Hi there, we have a problem with dahdi and overlapdial. We are running an E1 in Germany and are in need of overlapdial. The E1 is connected to a Sangoma A101. As soon as overlapdial is set to "yes" we have problems with incoming audio on the dahdi channels. When set to "no" all audio is fine. Basically we can choose between being able to receive calls or to place calls

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

I've got some weird behavior on my HFC-s cards. asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel 20040623 When i pick up my ISDN phone on Zap5-1 ("3987") and call the external number "1901" it will do so, connect me and everything is fine. In the second, where it tries to attempt the native bridge, the audio will disappear. Using another card