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
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:
Issue was that zapata.conf was specifying trunkgroup for the spans.
This was adding a DS1 Identifier to the setup channel ID, which
the CO does not recognize and so it assigns its own pick of channel.
This was causing a move which asterisk does not cope with.
(Actually, initial channel id being
in a PRI setup, the receiving side is changing the B channel at
proceeding.
Please post the layer 3 trace for this call scenario so that members of this
forum can help you debug your problem.
John Treble
Ottawa, Canada
-Original Message-
From: [EMAIL PROTECTED]
Here...
Please advise if any special flags/options are needed.
[ 02 01 01 27 ]
Supervisory frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
N(R): 019 P/F: 1
0 bytes of data
-- ACKing all packets from 18 to (but not including) 19
-- Since
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
If you mean the call keeps ringing, then the reason why is you have no
timeout in your dial statement. When there is no timeout the system
will never give up on line one so it can continue to the second
priority.
Thanks,
Steve
_
From: Joseph Rothstein [mailto:[EMAIL PROTECTED]
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
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
Hi Tim-
Except for maybe eliminating the channel bank as a secondary source of
timing, your conf files look fine. I notice that you've used 2 as the
build-out (line length) parameter - I take it that your asterisk box is not
right next to the switch. Anyway, I've never seen build-out make much
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
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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