similar to: dahdi and overlapdial problem

Hello, I have following problem with Sangoma A104 card: CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says

Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via PPP. Now, I want to freshen this setup to something newer. So I installed a Scientific Linux 6 server,

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

Im trying to install and configure sangoma ... every thing is OK but when type the command "wanrouter start" the following error apears: wan Driver not found. Thanks for any help

Hi All, I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box via BRI port in NT mode but actually I wasn't able to get it working. I've another standalone PBX, it's a Panasonic model, which works fine connected to the same port. The BT connected to a UK BT BRI line works quite fine after roughly a minute once connected. The PBX can be seen at

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

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root at asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device

Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10]

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 "wanpipe1 card 0" span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 "wanpipe2 card 1" span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI> == Using SIP RTP

Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us but we cannot hear the remote person? Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. WANPIPE Release: 3.4.1 Wanpipe Config:

Hi, I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years. The local telco finally got an ISDN switch and wants to move them to PRI. I didn't see this as a big problem - I've done a few others on this particular Caribbean island without issues, but this would be the first time with a Xorcom unit involved. We tried to do

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

hello, I have a asterisk box (Slackware 9.1.0, Linux 2.4.31) connected to a Ericsson Businessphone PBX on the internal S0 bus with HFC-S card and zaphfc driver - point2point mode -. --------- | TELCO | | BRI | --------- | | PBX external S0 -------- | PBX | -------- | PBX internal P2P S0 NT Mode | | HFC-S Card P2P TE Mode

Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on