Displaying 20 results from an estimated 2000 matches similar to: "Alphanumerical digits"
2003 Nov 25
10
PCI 3.3 V
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
Cristian VASILIU
AccessNET International S.A.
Software Programmer
mail to :<cvasiliu@accessnet.ro>
www:<http://cvasiliu.home.ro>
2003 Jul 11
7
ISDN PRI E1 configuration with E100P
<P>hi Everyone,</P>
<P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P>
<P>*CLI> <BR> == D-Channel on span 1 up<BR> -- B-channel 1 successfully restarted on span 1<BR> --
2003 Sep 23
1
Question about dialogic hardware
1. D/120JCT-LS card with 12 ports. This ports are FXS ports?
2. It is true that "Dialogic drivers cost of $15 per channel" ?
3. Can I use this hardware with asterisk (for E1-ISDN using Wildcard
E400P <http://www.digium.com/index.php?menu=wildcard_e400p>) ?
4. Anyone with experiance can tell me how they work and can provide a
configuration example? (2 DSP Motorola procesors - I
2004 Jun 28
1
Protocol Error (6) using Zaphfc
Hi!
Has anybody seen anything like this using zaphfc?
On outgoing calls (via isdn) , the line gets hung-up as soon as the called
party answers.
As seen below i get some protocol error (6) - but i'm not sure if this is
related to the "hang-up" which apparently comes a little earlier?!
Incomming calls on the isdn (zaphfc) interface is working just fine
(P.S. what about the
2010 Apr 10
2
PRI - Native ZAP bridge fails - Is this my patch?
Hi Guys,
I am calling out 416-999-1111 on Channel 1 of PRI and then calling
416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP
native bridged, both channels hangup and CLI show PRI cause (16).
Asterisk Verbose *(Channel 1 already connected to party)*:
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/4169992222
-- Zap/2-1 is proceeding passing it
2005 Oct 08
1
Outgoing call: hangup after answer
Hi,
When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get
immidiate hangup after answer. But when we place a full number before
dialing everything is ok. Any help appriciated!! Thanks
here is info with debug:
== Primary D-Channel on span 1 up
-- Executing Dial("SIP/200-164c", "zap/g1/|100|tc") in new stack
-- Making new call for cr 192
--
2003 Nov 20
2
Cannot do international dial with E1 in Spain
Hi,
I have a problem with dialling internationals numbers, and I don't now what
is the cause.
I have one asterisk with a e100p card connected to the Telco
(spain/telefonica) and it can dial local and national numbers without
problems but when I try to dial a international number it hangs-up. I call
the Telco to ask if the E1 can do international calls and it said that it
can.
I have tried
2004 Apr 18
4
PRI: This number has been disconnected
All,
When calling an invalid number using, I expect to hear:
"dooh-deeh-daah We're sorry you have reached a number which
has been disconnected ..."
And that is indeed what I hear when I dial out from [*]
using analog FXO, or VoicePulse or NuPhone. When I dial
that same number trough the T1 / PRI interface however, I
continually hear ringing, and then the call gets hungup.
Any ideas
2006 Jun 22
2
PRI Issue - Calls being rejected with unacceptable channel
Hey all. We have a DS3 circuit with GBLX split off into 7 systems with
a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P
cards in the other 5. GBLX numbers their spans from 0 to 3 instead of
1-4 and we have a NFAS configuration with the d-channel on chan 96. All
of our systems are running 1.0.7 for stability reasons (and no good time
for maintaince, the entire platform
2004 Apr 23
1
Busy error
Hi,
When have a incoming call from E1 to a extension FXS, and this extension is
busy, the incoming call recive ring tone, and it is wrong. What can I do?
Thanks in advance
Pedro
Here is the trace:
asterisk-1*CLI>
< Protocol Discriminator: Q.931 (8) len=41
< Call Ref: len= 2 (reference 66/0x42) (Originator)
< Message type: SETUP (5)
< Sending Complete (len= 4)
< Bearer
2006 Dec 02
3
Problem in Poland
Hello All,
I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing.
How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland?
Best Regards,
Alex
____________________________________________________________________________________
Do you Yahoo!?
Everyone is
2011 Jun 07
2
PRI issue its BUSY
Hi all,
I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect.
-- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is busy
-- Hungup
2004 May 09
1
No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier ----S2M------ * -----S2M------Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy
2006 Jan 16
2
Problem with calls starting from a legacy PBX
Hi,
I have this setup:
E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones
Can someone tell me what's wrong with this call initiating from an analog
phone connected to Alcatel PBX?
It dies with NOANSWER but all works if I call other destination numbers.
Dialplan is a simple Dial(zap/g1/0984465691) statement.
At the end you'll find also zapata.conf.
2005 Dec 21
4
[offtopic] Asterisk <-IP-> Siemens HiPath 4000
Hello!
Is it possible to connect Siemens HiPath 4000 to Asterisk? What
equipment required on Siemens side? I mean IP not E1.
Sorry for asking here. Siemens-related websites use "salesperson
language". There is no technical information.
2006 Jun 15
2
Bearer capabilities on PRI
Hey all,
I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware,
configured with a help from Sangoma Tech Support, running fine. It is
connected to a PRI circuit split from Cisco MC 3810, which in turn is
connected to a Converged T from CTC Communications.
While Asterisk works fine and I can call in/out on my BV account, I am
only able to dial in through CTC. I have spent
2004 Aug 29
1
not getting ringing/busy/answer feedback on my PRI
I posted a problem earlier thinking it was due to a lack of sound card.
Several members stated that you do not need a sound card to play audio to a
PRI channel. I did some further testing and discovered that there is a
problem with call progress tones or signaling on my PRI. I think that the
reason I am not hearing audio from the MeetMe() or Playback() apps. is
because the the calling side of
2007 Oct 31
4
PRI over T1 calls dropping, cause 100
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian
Option 61C. Calls either way drop with error "Channel 0/23, span 1 got
hangup, cause 100". Can anyone offer insight into the cause and
solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading
matching zaptel & libpri, put the problem is identical).
For testing, I tried a call from the
2003 Jul 28
2
"immediate=yes or Compleate recieved" with intcoming calls with new CVS
I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot
recieve the the calls from the zaptel interface which is a E100P with
pri signaling.
That is something with asterisk becouse rolling back to version from
06/23/03 using the new libpri and zaptel fixes the problem.
Here is an exept from the config:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension
2009 Mar 19
1
PRI QSIG Asterisk - Legacy PBX
I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0.
As EuroISDN it works fine.
However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why).
Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG.
So this