similar to: Early audio problem in chan_dahdi

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with,

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable

Some recent issues I had with hardware seem to come back to not understanding two very similarly named files: /etc/asterisk/dahdi-channels.conf /etc/asterisk/chan_dahdi.conf I've modified the chan_dahdi.conf to work now, but it would appear all I needed to do was include dahdi-channels.conf in chan_dahdi.conf and the problem would not have persisted? Is it me or is that a bit Monty Python?

Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP & IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI> module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550

Hello, I have a Sangoma A200 analog card with 2 FXO ports. It's working well with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk 1.6/dahdi. I compiled and installed, dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 asterisk 1.6.1.1 My analog card is recognized in dahdi_hardware. However, asterisk cannot compile chan_dahdi.so. I've tried passing

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load => chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed

Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and maintain an error-less Asterisk full file, with a single exception: I can't seem to remove `chan_dahdi.conf` without having Asterisk complain in the full logs:

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Hi, these are my first steps with DAHDI and I finally managed to get asterisk to load chan_dahdi (after I found out, that I need libpri). But how do I tell chan_dahdi on which isdn numbers it should react? I haven't found a parameter like "incomingmsn" for chan_capi in the documentation. Thanks for your help, Stefan

Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing even after they've been picked up... Here's one users summary: When I pick up the phone, I hear a dial tone and I am able to dial out. But for some odd reason, the receiving line picks up while the outgoing line is still ringing. And the receiving line can hear everything while the phone is still ringing. I tested

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hi, I am trying to build asterisk with xtensa compiler on embedded platform. I am trying to integrate my driver code to asterisk. For this tying to call driver code IOCTLs from chan_dahdi instead of dahdi IOCTLs. While compiling asterisk with xtensa, Observed chan_dahdi is not compiling [chan_dahdi.so is not creating].But with default settings [/usrc/src] it was creating. Please let me

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is

Hi All This may be a silly question but... I have this dahdi_genconf generated file: ; Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 4 22:05:29 2014 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI