I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows up and ok. This PRI is merely a crossover T1
going into an old DC0 class 5 switch.
I am getting the following errors over and over again
On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote:
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows up and ok. This PRI is merely a crossover T1
going into an old DC0 class 5
On Tue, Aug 16, 2011 at 10:48 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote:
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows
Eric Merkel wrote:
The error pretty much right away.
The first thing that comes to my mind is to check your cable.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Thanks to all who replied. The problem was due to a faulty NTU box from
the telco. It has been up for almost a week now without any downtime.
Regards,
Steve
Steven J. Douglas wrote:
Thanks for the tip, Harry. I will try that when I have exhausted all
avenue. My problem is that if I upgrade
Thanks for the tip, Harry. I will try that when I have exhausted all
avenue. My problem is that if I upgrade to 1.4.24 and DAHDI, I'll break
other stuffs.
In my current set up, the PRI did work for a long period of time (7
hours) before going into this unreliable mode (up and down). I'm
I had the exact same problem and errors some time ago (search the
archives for PRI dropping #2) using Asterisk 1.4.18, Zaptel and a
Digium TE121. I tried all kind of things, had telco technicians come
out and whatnot. The solution was two-folded - 1) I reinstalled my
server, 2) I updated to
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox D110P card on
asterisk version 1.4.21. It seems to me like a cable problem. I tried
using Ethernet straight cable (12, 45, 36, 78) and also a straight
cable where
Try a T1 crossover cable:
http://www.voip-info.org/wiki/view/crossover+T1+cable
On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.bizwrote:
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox
Hi Brandon,
When using the current straight cable, it sometimes worked i.e. I can
make calls from the PSTN into the asterisk. Do you still think that I
should try a crossover cable? Thanks.
Regards,
Steve.
Brandon B. wrote:
Try a T1 crossover cable:
Sent: May 24, 2007 7:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI problem, pri_fixup_principle: Call
specified,but not found?
Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding. It seems this sometimes breaks some logic
Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding. It seems this sometimes breaks some logic
(pri_fixup_principle) and then the hangup kind of breaks, release is not
answered and a restart cycle is triggered (by remote side).
Anyone can help me debug this ? I've seen
-users-
[EMAIL PROTECTED] On Behalf Of Carlos G Mendioroz
Sent: May 24, 2007 7:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI problem, pri_fixup_principle: Call
specified,but not found?
Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding
Mendioroz
Sent: May 24, 2007 7:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PRI problem, pri_fixup_principle: Call
specified,but not found?
Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding. It seems this sometimes breaks some logic
Thanks for the suggestion, but I cant seem to get
this to work for some reason.
When I dial my zap channel it does not seem to go beyond the
first priority.
I have setup the following just as a test, but never see the
output of Noop:
exten = _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
exten
On Sunday 08 January 2006 05:48, Joseph Rothstein wrote:
When I dial my zap channel it does not seem to go beyond the first
priority.
exten = _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
exten = _0.,2,Noop(${DIALSTATUS})
No application continues upon hangup unless there are special conditions which
]
Sent: Sunday, January 08, 2006 5:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] PRI problem
Thanks for the suggestion, but I can't seem to get this to work for some
reason.
When I dial my zap channel it does not seem to go beyond the first
priority.
I have setup
We have an Asterisk server with a single Digium E1. Everzthign works as it
should except for one minor issue.
When we place a call to a number that is busy, Asterisk does not seem to
properly send the busy signal back to the SIP phones. There is no indication
on the phone of anything at all, just
Looks like you got a configuration issue, you should test for the
${DIALSTATUS} variable and set the signalling to the phones based on
that.
You can do:
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Goto,s-${DIALSTATUS},1)
exten = s-CANCEL,1,Playtones(congestion)
exten = s-CANCEL,2,Congestion
currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet.here si the sip debug msg, we got a Message type: DISCONNECT (69) and
On Tue, Jul 12, 2005 at 10:59:42PM +0800, matt001 wrote:
currently we are able to use our USA sip phone to conenct into the E1 box,
but still unable to dial out to chinese phone numbers. They said from their
ISDN switch console, it shows D channel not connected to the voip server yet.
here
Tim, I would double check the timing. It seems odd that you would supply
clock rather than the switch, and if you get clock slips, that could
certainly account for what you are seeing. Feel free to contact me
off-list if you need more info or have any questions.
Bruce Komito
High Sierra
To: Timothy R. McKee
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PRI problem???
Tim, I would double check the timing. It seems odd that you would supply
clock rather than the switch, and if you get clock slips, that could
certainly account for what you are seeing. Feel free to contact me off
, May 24, 2004 08:30
To: Timothy R. McKee
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PRI problem???
Tim, I would double check the timing. It seems odd that you would supply
clock rather than the switch, and if you get clock slips, that could
certainly account for what you are seeing
On Mon, 2004-05-24 at 07:37, Timothy R. McKee wrote:
I thought the timing priority setting was for which incoming timing signal
was used as the primary clock source, so I set the PRI as the highest
priority clock source. In the telco world this is that way it normally
works. Does the
PROTECTED] On Behalf Of Timothy R.
McKee
Sent: Sunday, May 23, 2004 7:52 PM
To: [EMAIL PROTECTED]
Subject:[Asterisk-Users] PRI problem???
I have just finished installing a new asterisk box at my work. The box is
quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel
Sent: Monday, May 24, 2004 12:58
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI problem???
Hi Tim-
Except for maybe eliminating the channel bank as a secondary source of
timing, your conf files look
I have just finished installing a new asterisk box at my work. The box is
quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4
port Digium T1 card for channel bank and PRI access.
I activated a PRI from a local CLEC (DMS-500 based, National protocol).
This PRI is on slot 2
On Sun, 2004-05-23 at 21:52, Timothy R. McKee wrote:
My problem lies in random intermittent drops of calls. The entire PRI seems
to disappear, dropping all current established calls. I see occasional
printouts on an asterisk management console showing all 23 B channels
resetting with no
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