similar to: No translator path exists for channel type DAHDI (native 76) to 256

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 |

Hello, I'm experiencing some weird problems on my server: - 1) The following messages are filling up my logs: [Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Sep 29 08:24:59] WARNING[7078]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 171 as D-channel anyway!

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can

Dear Rxperts! A simple example where "write.csv" does not seem to accept user specified arguments.. Why? write.csv(t(1:10),"./te1.csv",quo=F,col.names=F) Warning message: In write.csv(t(1:10), "./te1.csv", quo = F, col.names = F) : attempt to set 'col.names' ignored However, write.table does fine..

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:

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

I've done a decent size update of the OpenSSH portable from the upstream tree. There is an AES upgrade that needs to take place, but I need people to test and tell me what endedness changes need to be applied (if any). Attached is the patch. It is geared towards the latter snapshots..Unsure how it will patch (if at all) against 2.9pX series. Thanks. - Ben "This is dark day in

Hi there! I am having some difficult in receiving calls from my E1 link using mfcr2. I can make calls normally , but when I receive an incoming calls, the phone ring I answer it ,so, I listen busy tone and then the phone ring again and again. look the log: -- Executing [4801 at from-pstn-TE1:1] NoOp("DAHDI/2-1", ""1233220567" <1233220567>") in new stack

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

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 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 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

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

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE!

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

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I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

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 am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24