similar to: Can't make call with TDM410P

Thanks I had this line in my /etc/asterisk/chan_dahdi.conf : include=/etc/asterisk/dahdi-channels.conf the file /etc/asterisk/dahdi-channels.conf was generated by /usr/sbin/dahdi_genconf I simply did that : cat /etc/asterisk/dahdi-channels.conf >> /etc/asterisk/chan_dahdi.conf It works now. May be the option "include" is not supported within the file chan_dahdi.conf

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hello, I have just installed an Asterisk server with a Digium TDM410P card with 3 FXO modules (no module in the 4th slot). It's lived on two different machines (a test machine, which had Linux kernel 2.6.28, and a new dedicated machine which has Linux kernel 2.6.32). On the test machine (2.6.28), I used the Zaptel drivers. Once the kernel modules were loaded, the lights on the TDM410P came

Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 * *compiling Asterisk-Cert2 1.8.11* *./configure make make install make config * Afther that i register the DPMA license, and finally copied the * res_digium_phone.so* to

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver? Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

Hello, I am installing dahdi on a machine lspci 00:00.0 Host bridge: Intel Corporation 3200/3210 Chipset DRAM Controller (rev 01) 00:01.0 PCI bridge: Intel Corporation 3200/3210 Chipset Host-Primary PCI Express Bridge (rev 01) 00:06.0 PCI bridge: Intel Corporation 3210 Chipset Host-Secondary PCI Express Bridge (rev 01) 00:1a.0 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of

As recommened I got the new firmware for my echo cancellers and it solved hte problem with the agressive echo cancelling causing half duplex audio. I have to say, so far these cards are far superior to the previous models. The sound quality is hugely improved (enough to really notice which is alot) and the echo canceller works way better than the software ones. My system seems to like these

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

Hi All, Running asterisk 11.25.3, the /proc/cpuinfo says Intel(R) Xeon(R) CPU X5675 @ 3.07GHz lsmod | grep dahdi gives dahdi_transcode 16384 1 wctc4xxp dahdi_voicebus 61440 2 wctdm24xxp,wcte12xp dahdi 225280 11 wctdm24xxp,wcfxo,wctdm,dahdi_transcode,oct612x dahdi_voicebus,wcb4xxp,wct1xxp,wct4xxp,wcte11xp,wcte12xp crc_ccitt 16384 2

Have a new TE133 in a dell R320. dahdi_cfg -vvv shows: DAHDI Tools Version - 2.9.1 DAHDI Version: 2.9.1 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear

Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root at pbx ~]# lsmod | grep -E "hisax|netjet|dahdi" netjet 14618 0 isdnhdlc 4523 1 netjet mISDNipac

On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327