Displaying 20 results from an estimated 500 matches similar to: "DISA restart from begining"
2009 Mar 24
4
PRI dropping
Hello,
I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo
cancellation. Every 30-60 minutes I experience PRI dropping.
@@@ /etc/zaptel.conf:
loadzone=dk
defaultzone=dk
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
@@@
@@@ /etc/asterisk/zapata.conf
[channels]
switchtype=euroisdn
usecallerid=yes
group=2
signalling=pri_cpe
context=incoming
channel => 32-46
2009 Mar 26
2
PRI dropping #2
Hey,
I wrote yesterday about PRI dropping, which turned out to just be a
regular reset of unused B-channels. This time there's a real issue. As
noted earlier I have an ISDN-30 connection, a Digium TE-121 with
VPMADT032 echo cancellation. These are my configurations files:
== /etc/zaptel.conf
loadzone=dk
defaultzone=dk
span=1,1,0,css,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
==
==
2004 Jan 12
1
E100P - connected to Cisco
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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<head>
<meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
Hi,<br>
<br>
have any one sucessfully connected E100P to Cisco? I am
2003 Aug 18
8
PRI Question
I managed to get Asterisk working with my PBX using T1, now I am moving
on to trying to make PRI work.
I have my zaptel.conf and zapata.conf configured as follows:
Zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
Zapata.conf:
[channels]
transfer=yes
immediate=yes
callprogress=yes
language=en
context=default
switchtype=national
signalling=pri_net
group=1
problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)
2006 Feb 17
2
problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)
Nik,
This definitely helps! Please check your dial command. You've got
"Dial(Zap/0/mynumber)" and I think you might possibly want it to be
something like this:
Dial(Zap/1/mynumber) or
Dial(Zap/g0/mynumber)
I don't recall there being a zap channel zero, but it is common to have
a group zero. I would recommend trying Zap channel 1 -
Dial(Zap/1/mynumber) - before trying the
2007 Oct 02
5
PRI Setup problem
Hi everyone,
I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI
(Megalink) circuit and having some trouble getting it to handshake.
Thanks for any help or suggestions to diagnose this problem.
The problem is that Asterisk has trouble bringing up the link. I see
the following lines every couple of minutes:
== Primary D-Channel on span 1 up
== Primary D-Channel on
2004 Jun 15
5
PRI problems (telewest -> * -> LG GDK 186)
Hi,
?
I'm trying to figure out what the issue is splicing Asterisk between our
Telewest PRI and a GDK-186 with a PRI card.
?
We're using the Digium TE405P
?
Our telco provider is Telewest, and Telco directly into switch is fine.
?
When I splice Asterisk in, I can make and receive calls from Asterisk
extensions, I can make outbound calls from the GDK, but inbound calls do not
seem to pass
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.
2008 Dec 08
2
PRI span debug out put - failing international calls
I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error.
Kind Regards:
Gabriel
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2005 Oct 03
1
Direct Dial In - second try
Hi all,
I have an asterisk-server (cvs-head from august) connected to a
carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems
with DDI (standard 'official pstn' number plus extra digits for
'internal' use)
Basically, when the entire number (including the extra digits) is
dialled via a redial or a programmed key, I see the entire called party
number (including
2008 Feb 19
1
A problem about digium TE220B
hello everyone,
I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected
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
2006 Apr 08
1
ANI on a PRI
Is there a setting somewhere in * to define whether I am receiving
callerID or true ANI? Global Crossing claims they are sending ANI but I
dont think so. My understanding of ANI is that it is always sent,
regardless if callerID is blocked. If I dial *67 and my DID, I get
"Presentation: Presentation prohibited of network provided number" and
no number.
Before I call GC on Monday
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
2004 Aug 02
2
CallPres screening DDI
Hello,
we had a running configruation where asterisk passed the phone number and
the ddi to the pstn (i.e. 595-431)
Now only the rootnumber arrives: 5950
I do not know, what to do. I tried to use callingpres (now i am just hiding
every number, because 595-0 is no valid extension..) but that did not
worked.
> Protocol Discriminator: Q.931 (8) len=44
> Call Ref: len= 2 (reference
2007 Dec 06
2
Print CALLERID in CLI during "pri debug "
Hi all,
I was wondering if it is possible to print the callerid value in the
CLI when doing 'pri debug span 1'
For example
> Call Ref: len= 2 (reference 2707/0xA93) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 97]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0
> ChanSel: Reserved
>
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
2006 Mar 22
3
PRI DMS100 -> Nortel Meridian Option 81
Hello all,
I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option
81C system. The PRI line is currently setup as DMS100. Here are the
relevant lines from zaptel.conf and zapata.conf:
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone = us
zapata.conf:
[channels]
language=en
context=from-internal
musiconhold=default
switchtype=dms100
2004 Sep 16
2
No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten => s,1,SetCIDName(Test)
exten => s,2,SetCallerID(1234561234)