Displaying 13 results from an estimated 13 matches similar to: "dahdi_test drops after restarting Sangoma driver"
2008 Oct 06
2
Conneting Asterisk to Swyx pri
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
2010 Oct 01
2
No translator path exists for channel type DAHDI (native 76) to 256
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
2009 Dec 03
2
dahdi_tool shows no alarms, but no line connected
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 |
2010 Jul 02
1
asterisk and cisco 2800
Hi all,
I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures
with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the
cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives
no errros, the span is up and active, green light on the card) but when
I make a test with my iax phone, there's no way to dial the PBX and I
get this WARNING:
[Jul 2
2006 Mar 01
3
my zap channel not ringing
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
2009 Oct 23
1
Strange IAX2 / Iaxmodem problem
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!
2014 May 12
1
Terrible dahdi_test results
Hello, I am trying to get a Wildcard TE110P to work in a relatively
modern HP Proliant DL385p Gen8 server. Being a potent 12 core Opteron
server I expected no problems.
Much to my dismay the dahdi_test results are constantly terrible:
# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
89.101% 89.195% 89.142% 88.957% 88.953% 89.115% 89.089% 89.134%
89.066% 89.021% 88.933% 89.044%
2007 Jan 03
2
Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
I've replaced 2XTE110 with an A102 with echo cancellation specifically to
deal with echo problems. However, user feedback has indicated to me that on
some calls (not a lot, but some) the call is unusable, with audio
artifiacts, described by one user, as: "very bad phasing reverb & feedback
(from my rock & roll days)". This is quite intermittent, as in most cases,
the user
2010 Oct 26
3
Channel Bank ? Simple Switch Hangup?
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
2010 Sep 15
1
One way audio when overlapdial is set to 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
2009 Feb 04
3
siemens hipath 4000
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
2008 Oct 21
1
hex b1 in CallerID sent by Asterisk On PRI
I'm trying to send CallerID info to a MetaSwitch system over a PRI. The
MetaSwitch gets the info exactly as it is sent by Asterisk, but I think
it might be having trouble with the Hexadecimal b1 that is being sent
just before the first character of the CallerID Name.
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1
2006 Mar 02
0
RE: Asterisk-Users Digest, Vol 20, Issue 13
On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote:
> Does anyone have a way to do wake calls?
>
>
>
> Jordan Novak
>
> Communications Technician
>
> Logistics Health Inc.
You could use cron and /var/spool/asterisk/outgoing scripts to dial
numbers, etc...
>
Can you elaborate, I am fairly new to Linux and a phone guy to boot. I
am looking for a way for the