similar to: cannot get whole DNID with ISDN line

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

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten

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

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. ->

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

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

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

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

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

I'm running a setup with chan_misdn on a austrian PTP-line. When somebody dials in without an extension, he gets a busy signal. I don't see the call at all in asterisk. I *have* set immediate=yes in misdn.conf. And I *do* have an s-extension in my dialplan for the context used by misdn. Calls that provide an extension work fine. Attached is my misdn.conf and a verbose 3, misdn set debug

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

Have you got termination set correctly? I have a B410P working with 2 x NT and 2 x TE ports successfully. I had to turn the 100ohm termination on on the NT ports (even though I have them set as PTP in mISDN.conf). HTH -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On

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

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:

Hi, Does anyone has a *working* patch to set the dnid from extension when it is now supplied, cos this plugin doesn't working without it. I tried and it passes empty string. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/aae0a7f4/attachment.htm

I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?

hello i am using ast-rad-acc.pl from portaone connected with asterisk manager. my (%cdr) = @_; $cdr{'CALLERID'}, $cdr{'DNID'}, these are empty why these two variables are not working on hangup any comments thanks Kamran Ahamd __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi, I installed asterisk 1.2.1 on a debian distro with 2 tdm400p cards. I'm receiving the DNID instead of callerid...this is very strange. I found many errors and warnings inside asterisk log like these: Feb 28 15:28:12 ERROR[18218] callerid.c: fsk_serie made mylen < 0 (-14) Feb 28 15:28:12 WARNING[18218] chan_zap.c: CallerID feed failed: Success Feb 28 15:28:12 WARNING[18218] chan_zap.c: