similar to: ISDN -> SIP

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_

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----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 12:36 -->> To: asterisk-users at lists.digium.com -->>

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 i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm:

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 everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi, i use an BN4S0 with misdn an asterisk on Linux 2.6.9. The hfcmulti module is loaded with option: type=0x04 protocol=0x2,0x2,0x22,0x2 layermask=0xf,0xf,0xf,0xf and the fourth port is connected to an ISDN PTMP (MSN) port. Call to #72 from S0 (BN port 4) are not accepted from asterisk but why ? Can anyone give me a hint ?? misdn debug messages follows: lib: NEW_CR Ind with l3id:80001

Hi, Since a week ago I am trying to get chan_misdn working with asterisk 1.4.19, using HFC based ISDN card on Linux 2.6.22. My setup is done as detailed on wiki and FAQ. * mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed. After installation and misdn-init, I have this: aragorn:root/pts/1: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) ->

Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet -->

Hi. I want to handle incoming chan_misdn traffic by asterisk, but I've got message - 'Extension can never match, so disconnecting'. What I'm doing wrong ? How I can pass incoming dialed number (dad) to misdn context (in my case 'dss1_incoming') ? Works unrouted calls (s extension) if I set immediate=yes in misdn.conf, but I want to route calls by dialed number. log

Hi I try to connect my asterisk with a Siemens Hicom pbx. I have a PCI cologne Chip card wich support NT mode. I have compiled mISDN driver, and I use chn_misdn from debian package. The card wotk fine in TE mode but mot in NT mode. for informations : routeur*CLI> misdn show stacks BEGIN STACK_LIST: * Stack Addr: Uid 40200001 Port 1 Type NT Prot. PMP Link DOWN --> bchan: addr 0 channel

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

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

_____ Von: Pascal Speck [mailto:p.speck@ewersbach.net] Gesendet: Montag, 5. September 2005 10:37 An: 'asterisk-users@lists.digium.org' Betreff: Timeout when Dialing - HELP When i try to do a call I get this message after a few seconds: I IND :TIMEOUT pid:1 mode:NT addr:51400102 port:2 --> l3id:10040 cause:16 dad:800759 oad:20 channel:1 port:2 --> lib: prim 34582

Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the "dad" is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ?146472130 dad: ?146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can

hello everyone! I just got the newest asterisk SVN: trunk# svnversion 152803 and compiled it. then I made some test-calls. 1. Calling my mailbox. It worked, but quality was not good, in comparison to 1.6.0-beta9. I called via mISDn. 2. Just call myself. Result: Ringing and asterisk segfaulting. 3. Same for calling some gtalk-number and using app Dial mISDN/1/my_number. I got this a