similar to: incoming DAHDI Channel explained

Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a "Dial(DAHDI/g1/voicenumber,r)". The call is bridged and everything is fine. "dahdi show

Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in "/dev/zap"(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : )

Here's my planed setup : PRI from telco <--> (port 1 of A104d) * (port 2 of A104d) <--> PM3 The PM3, for those who don't know is lucent's portmaster RAS dial-up router. I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards). In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2 is TDM_VOICE, without hw echo, Clock as

Hi, Recommending to go Digium because of an OpenBSD issue with the Sangoma A104D is quite funny to say the least since neither is the TE407P supported in BSD by Digium. So the recommendation is useless to the person who originally requested for a comparison between the two products. If someone wanted to send back the A104D, he could have taken advantage of Sangoma's 30 day money back

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

Hi there, I am running Asterisk 11.9.0 WANPIPE Release: 7.0.10 DAHDI Version: 2.9.0 Echo Canceller: HWEC libpri version: 1.4.12 When I start the ConfBridge application I get the following warning: [2014-10-24 14:36:21] WARNING[29177][C-00006934]: config_options.c:790 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) I do not specify a specific

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

Hi, I am facing an audio-problem with the dial application and I (!) think, that it is connected to the dahdi parameter "overlapdial=yes". Sangoma support does not see any connection between this. But when enabling this option I face with some(!) dial-partners a audio one-way issue (the called party can not be heard). Only using PSTNS (germany e1 trunk) - no voip. Is there any

I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

Hi everyone, I am reading through Sangoma Wiki right now. But someone may already and quickly notice this. I have a system that is down since the morning (maybe power intruptions). All seems fine except for "wanrouter status" shows disconnected. Following are the alarms raised. Should I call telco (they have long wait times) or should I just keep searching online for troubleshooting

Howdy, Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel, zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source, asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source, Sangoma A104d with firmware that is probably a year old). I plugged in an RBS T1, ESF, B8ZS, wink start, and MF signalling. I stuck with the defaults that the wanpipe build

Hello, I've recently bought a new Sangoma A104d PCI-card. WANPIPE release is 3.4.7. Machine is a dual Xeon with Debian 5.0.3. Asterisk is 1.6.11 with recent libpri and dahdi. When I boot the machine (including hardware and wanpipe and dahdi drivers) and start asterisk, everthing runs fine for almost 5 minutes. Then every few seconds HDLC errors occour. I also tried booting - and starting

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

Hello guys, I have trunk ?10000", Internal num ?99" and MeetMe ?1010" now I calling 99 -> 89264959635 via 10000 /pbx/agi.php: [agi_channel] => PJSIP/99-00000012 /pbx/agi.php: [agi_callerid] => 99 /pbx/agi.php: [agi_calleridname] => 99 /pbx/agi.php: [agi_context] => dialmap /pbx/agi.php:

Hi, On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio. My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx "pri show version" libpri version: 1.4.12 A

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value