similar to: DTMF tones mid conversation

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,

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

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:

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

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:

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] -->

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 helpers, I seem to have a problem of intermittent DTMF tones being played during a conversation. Eg: Extn 100 takes an inbound call and all is fine. Except, at an undetermined time the person on extn 100 will here a DTMF tone for no apparent reason (it's not the caller pressing buttons). The caller doesn't hear the tone - only the called person. The call itself progresses normally.

Hi, I've been able to make make install the Zaptel drivers (1.2). I'm using a b410p so I executed the following command make b410p. I tried this on multiple machines, but it always failes: root@asterisk:/usr/src/zaptel-1.2.11# make b410p [ -f misdn-b410p.tar.bz ] || wget ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz --23:59:54--

Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz

Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1

Hi list, I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in zaptel dir and: make, make install, make b410p. Everything is ok. Now I want to compile Asterisk but can't activate the chan_misdn channel which depends on -from menuselect- isdnnet(E), misdn(E), suppserv(E) When I made the make b410p, all

Hi, I'm using a 4-port BRI card (b410p) to make and receive calls (via chan_misdn): http://www.digium.com/en/products/digital/b410p.php This card supposedly has hardware echo cancellation. How can I check that echo cancellation is actually ON (taps, etc) on a given misdn channel (like with the "zap show channel X" command). "misdn show channel X" doesn't seem to show

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

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

Hi, I have two Asterisk systems that hang once every 4-6 days (more or less). One has * 1.2.31.1 and the other 1.4.26.2. The last system collapsed today and I saw several messages looping endlessly on screen: hfcmulti_rx no memory for rx_skb alloc_stack_skb(303,110): no skb size During this time the server is completely down (no ping). A forced reboot is necessary (very ugly situation in

The DAHDI docs actually include the 'steps' that used to be in Zaptel's 'make b410p'. These steps involve downloading and compiling mISDN. Why re-invent the wheel? Just a thought :-). As for the 100ohm termination bit - it's simply changing a couple of dip switches on the b410p card (as described in the manual). Andy -----Original Message----- From:

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn