similar to: misdn, no free channels, similar to FAQ one

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'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 am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten => 12345,1,Dial(SIP/012346737222 at sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after

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 -->>

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:

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 all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick up the phone. If the destination phone is picked up, then everything is allright and the

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

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 -->

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

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] -->

Hello list, I've been having a problem for some time now that I can't figure out how to solve it. On a PTP BRI ISDN line, if I have both channels in use and I place a third call from the outside, I'm not getting a busy tone like I should. Instead I get a congestion tone, as if the line was not working/didn't exist. Currently I'm using mISDN 1_1_9.1 and Asterisk 1.4.35 with

I hadn't used FWD for quite a while. A customer sent me an email last week, "Is FWD broken when one tries to use it with IAX?" I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

Hi all, I'm trying to get a digium B410P to work with asterisk 1.6.0.1 (but i have the same problems with asterisk 1.6.0) Official digium documentation cover up to asterisk 1.4.x and suggest to use zaphfc... but as i understood zap is completely gone in 1.6 I tried with misdn (1.1.8, 1.1.7.2 and 1.1.5) and card is recognized correctly by the kernel but chan_misdn does not work as expected.

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

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

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

Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0

Its been a few days, I was wondering if anyone else has any ideas on how to get this to work? If not, could I ask a the direct question as to if anyone here has (or knows someone who has) successfully got something like a Cisco Router to successfully establish an ISDN data/Internet connection through the Asterisk PBX?