similar to: problem with mISDN

Displaying 20 results from an estimated 500 matches similar to: "problem with mISDN"

2007 Jun 13
2
mISDN problem
Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is
2009 Mar 12
2
BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai
Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0
2007 Jan 14
1
Problems with mISDN TE line
Hi list, I've installed Asterisk 1.4.0 with newest mISDN 1.0.4 + mISDNuser 1.0.3 on Fedora Core 6. I get many compilation error on mISDN. It wants to include linux/config.h That I fixed by removing the #include line at every occurance. (Don't know if that was a wise move, but it then compiled). mISDNuser and asterisk compiled fine, and asterisk can find and use the ISDN BRI port in
2009 Jan 05
1
B410p, Ast1.4, France Télecom Numeris Double T0 problem
Hi. When I call my RNIS numbers (with a mobile phone for example), I can see 2 incoming calls on the IPBX, which should not happend. I'm not sure if it's a problem with the telco France Telecom and their ISDN setup, or if it's a problem with the MISDN driver on the IPBX itself. I'm stuck ... Any advices for troubleshooting that? Someone provide working configuration files
2009 Sep 08
0
Intermittent metallic voice SIP->ISDN ISDN<-SIP
Hi all, I'm fighting with a really strange problem that is really busting me. I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7 3 extension on hardphone and 3 extension in softphone ( zoiper ) What happens is that sometimes the people on the other side of communication hear my voice as metallic and chopped. This happen either on incoming call than on outgoing call. If I
2007 May 08
0
Beronet card - issue?
Hi all, I have a problem with my beronet card with 2 isdn. I think drivers and Asterisk are ok but the red led on the card always blinking. The card is connected with PBX. I post some conf: [root@gateway ~]# misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib. ->
2007 Jul 24
0
mISDN & Asterisk 1.4: HFC-S card not responsive
Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk
2006 Dec 28
1
mIDN question
Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT
2007 Jun 12
1
call from ISDN
Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new
2009 Nov 05
2
faxes received on mISDN
Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as
2009 Jul 20
0
No subject
I'm wondering if hdlc can be the culprit (not sure what it is and what it does). Should I set hdlc to yes in misdn.conf (I'm asking before testing because this is a production system)? misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ntkeepcalls=no bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes
2009 Feb 06
1
set caller id on outgoing calls through BRIISDNlines
You're quite right. We'll need to see your misdn.conf file to check the settings. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 13:49 -->> To: asterisk-users at lists.digium.com -->> Subject:
2007 May 08
2
outgoing calls
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: zaptel.conf Type: application/octet-stream Size: 34 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/f4c36cab/zaptel.obj -------------- next part -------------- A non-text attachment was scrubbed... Name:
2007 Dec 15
0
OpenVox B800P and asterisk 1.4/ mISDN-1_1_7
Hi i've installed this software: ******************** SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 ******************** SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) ******************** MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no
2007 Apr 26
0
problem with A400P01 OpenVox
Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
2007 Apr 27
1
can´t anserd the call
hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:
2009 Sep 01
0
mISDN NT mode config setting
Hi, I am struggling to get plain Cologne chip cards to run in NT mode, runs nice in TE mode despite the error message: login as: root root at 192.168.2.22's password: Last login: Tue Sep 1 23:09:24 2009 from 192.168.2.50 Welcome to Elastix ---------------------------------------------------- misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1
2007 Nov 17
1
Building and running mISDN for B410P on Ubuntu 7.04
Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using "make b410p" I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using "make b410P" on Ubuntu? (make force does not help at all) 2) In some of our installations
2008 Nov 05
0
b410p mIDSN - RNIS signaling problems
Hi. I'm running Asterisk server with 10 sip phones, and 2 grouped T0 lines with 10 DDI numbers. My provider is France Telecom and my setup is : - Debian Lenny - Asterisk 1.4 - Linux kernel 2.6.25.17 - mISDN 1.1.8 driver - Sip phones Thomson ST2030 No problem with the SIP . But when reveiving a call on RNIS line (any of the DDI numbers), the associated SIP phone rings indicating _two_
2007 May 09
3
select menu
Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>