similar to: no extension from ISDN phone with bristuff

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

hello: thanks for Tzafrir Cohen for dahdi testing. I installed dahdi-2.1-r3c svn code and asterisk1-6 for testing OpenVox B400P and junghans card. i fund that there is bug (i think) to dectect NT or TE mode. actually on the board, i set it as TE mode, but after start wcb4xxp, but it show the port is NT mode. to detect the TE mode, I modefy the code in base.c

Hello gang, We just got MaBell to turn on our callerid. I tested the capability with a southwest bell box and a plain phone, so I know the line is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a TDM400P card. Here is my dahdi_cfg -vv output: dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.3 Echo Canceller(s): MG2

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

> On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: >> Only way I have managed to get Zap channel to reject a call on TE110P >> without answering it, is to dial number that is not handled in dialplan >> (I >> have a ISDN PRI with 100 number DID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >>

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Hi, When startin * it tells me: ================================ [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, PRI Signalling signalling Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf:

I've been beating my head a bit against the 1.0.6 Debian builds of Asterisk, using an E100P (E1, single span) board. In machines I've built in the past (back in 1.0.0 days), config I'm using and that card and 1.0.0 driver combo worked fine. ztcfg reports no problems: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. And zttool sees the card, and

I am running * with a Junghanns quadbri that should allow us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the

so here it is, the problem that's been nagging me for the past 2 days: connected a box to my telco's NTBA <-> zap/asterisk. which works: box:/etc/asterisk# cat /proc/zaptel/1 Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In

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

I am trying to retrieve the cause code of a outgoing call over a PRI where the number called is out of service. When an out service number is called I get a recording that the number dialed is not a working number. I see cause code 1 in in the CLI as soon as the call is dialed the Telco recording goes on for 30 sec. then hangs up. Any idea on how retrieve info that the called number is is

Hi there, I just upgraded a relatively old Asterisk installation (1.2) in our office to a relatively new version (1.6svn from last wednesday) which runs a Junghans QuadBRI card [1]. To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as well. After a while of juggling it "works". What doesn't work: connected ISDN devices (Gigaset phones connected to QuadBRI

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national