Displaying 20 results from an estimated 2000 matches similar to: "PRI CallerID - leading zero added"
2007 Sep 26
1
Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
Hello,
Digium support kindly proposed to ship a TE120P card to help resolve the
issue.
I plugged in the card, and introduced the loopback plug. I cleared the
red alarm for a while and then i started seeing alarm switching from
Yel/Recovering to Blue/Rec with a lot of IRQ Misses.
I call Digium that assisted us, and we noticed IRQ sharing with the VGA
adapter and the Ethernet port.
I changed
2008 Feb 25
1
DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After a call placed any DTMF tone causes the server
to lock up entirely. Calls placed work just fine (except for a problem with
echo cancellation). The phone registered to the server is a Linksys SPA-942.
I am seeing in zttool some IRQ misses, but it never seems to go above 74
(below).
2008 Jan 26
1
CHANUNAVAIL
I've got a setup where we have 100 DID's. Our default dialplan has one
line that calls a macro:
exten => _22XX,1,Macro(STDEXT,${EXTEN})
The macro is pretty basic:
[macro-STDEXT]
exten => s,1,NoOp
exten => s,2,Dial(SIP/${ARG1},15,Tt)
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1}|u)
exten => s-NOANSWER,n,Hangup
exten =>
2005 Feb 12
0
Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think the root cause is that DIALSTATUS gets reported
as BUSY. The debug output is below. My desired
2007 Sep 20
9
Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
I am trying to connect two machines to each other with an T1 crossover
cable. The first machine has two TE120P cards - one connecting to the telco
on an ISDN PRI. The second to a crossover T1 cable to a second machine which
has one TE120P card.
Telco <-cA-> Machine1 <-cB-> Machine2
Machine1: Two TE120P cards
Machine2: One TE120P card
cA: Standard T1 Cable
cB: Crossover T1
2006 Apr 25
1
CHANUNAVAIL, busy and congestion
Greetings to all,
I ma having a problem with channel variables on a couple of our Asterisk
boxes.
Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our
external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN.
On the External GW, we also have an IAX trunk to a VOIP provider if for some
reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be
2010 Sep 23
1
Net2Phone SIP trunk problem
Dear, I have this scenario:
- PBX Asterisk 1.6.2.10 with private IP 192.168.0.10
- Behind a Cisco ASA firewall that connects to Internet
- SIP trunk to Net2Phone with these parameters (nat=no):
host=200.58.113.60
username=DOLLY
secret=123456
port=5060
type=peer
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw
nat=no
canreinvite=no
qualify=yes
-Softphones Xlite
The PBX can't register to
2009 Oct 07
2
Can dial long distance but not local?
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I'm sure I just have something goofed up in the
dialplans? I have a bunch of Polycom 331 IP phones connecting to the
server. I can dial the other extensions in the system fine and I can
dial long distance outgoing but cannot seem to get it to dial local (7
digit) calls.
I see this in the CLI:
--
2008 Jan 29
1
PRI Alarms, Comes Back, But Asterisk Won't Touch It!
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
(same problem with various previous versions; same problem with
different TE120P cards).
The customer has a partial (10 B-Channel) PRI that when it is busy
(eight or more B channels in use), tends to fail as shown below...
[Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown
error 500
[Jan 26 23:00:31]
2009 Sep 15
0
1.6.2.0-rc1 intermittent voicemail problem ?
1.6.2.0-rc1. I am having trouble with voice mail intermittently not
working correctly on CHANUNAVAIL. (it may happen for other statuses
too, haven't checked). Basically here's what happens:
-- Executing [1651xxxxxx at mydids:1]
Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack
-- Executing [s at macro-phone:1]
2009 Sep 22
2
Problem with dialplan -> gotoif ?
Hi
This is the output from show dialplan dial-sipmnf-sippt-pstn
[ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ]
's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config]
2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config]
3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
2007 Aug 13
2
Does Digium TE120P card support MFCR2
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
2007 Nov 28
1
Digium TE120P versus Sangoma A101D-X
Hello List,
We purchased a TE120P card from Digium and it works great. The only
problem is that we are still experiencing echo on some calls. I've tried
various echo cancellers (right now we are using OSLEC) and still no
luck.
My question has anyone gone from the TE120P to a Sangoma A101D-X Single
Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference?
Also I called
2006 Mar 06
0
No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer
2004 Nov 30
0
No voice when I dial out
I can dial from 601 to a public number.
The public number rings. I pickup and hear nothing, while on 601 it keeps ringing.
(BTW, is it right to say "ringing" on the active phone?)
The *CLI> doesn't show me anything useful:
Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack
Executing SetGlobalVar("SIP/601-8238",
2007 Aug 13
1
Does digium TE120P card support for MFC/R2 protocol
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
2007 Apr 02
1
TE120P and Unknown Signalling Method
I have a brand new TE120P card that I have installed and asterisk is not
starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown
signalling method 'pri_cpe'
It seems it does not matter what I change the vaule for signalling= to,
it always returns it as invalid.
I have tried the config from my other 2 servers running TE110P cards and
the config from AusTechPartniships
2007 Dec 09
1
Installing/configuring TE120P debian way
Hi all
I use asterisk (1.2 brach) from debian official packages and it works fine.
Now I need to install and configure a Digium TE120P card, but I cannot
find any guide to install it using debian packages.
I would like to know if anyone of you knows about packages that would
include the necessary kernel modules or any other method that won't be
broken when the asterisk packages are updated.
2007 Oct 17
3
Asterisk using 200% CPU and then crashing...
We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1,
Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual
Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two
TE120P cards and everything was working fine. Since they needed to add
a third E1 line we decided to change one of the TE120P cards with a
TE210P. After the change we had a couple of crashes (server
2007 Feb 09
1
Outbound Call Transfer Problem
Hi
I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.
The problem happens:
- With both software and hardware phones.
- With calls going out through the ZAP channel and to internal SIP
extensions.
- After I have transferred an