Displaying 20 results from an estimated 8000 matches similar to: "change E1 link from ISDN to Q.SIG"
2006 May 12
1
TE110P on E1
Hi,
I wonder if anyone is using Digium's TE110P card on an E1 connection.
I have been try to, but so far it wasn't much of a success.
It only works more or less in EuroISDN as PRI CPE.
And even that config gives me some trouble with channel negotiation.
My current config:
*zaptel.conf:*
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=be
defaultzone=be
*zapata.conf:*
[trunkgroups]
2009 Mar 13
1
Asterisk to Ericsson MD110 on E1 with ISDN-USR (not QSIG)?
I have been asked by a potential customer whether we can connect an
Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR.
They are unable or unwilling to upgrade their E1 port to QSIG.
Has anyone here had experience of successfully making such a connection?
I have found a couple of hits on Google that suggest it "should" work,
but I'm after something a little more
2004 Apr 30
1
Configuring Digium TE405P for use in Germany
Hello all,
I really checked voip-info.org but it still seems to be not very easy and I
just hope that there is anybody with a simular config.
We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second
PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper
than a channel bank :-)
Carrier ----S2M------ * -----S2M------Siemens
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2005 Sep 13
1
Integration between Asterisk and Siemens HiCom 150e over ISDN
Hi,
I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and
wondered if anyone is able to offer any advice.
In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk
box. e.g: HiCom user dials access code and can call Asterisk extension or
establish SIP call over Internet. Likewise, I'd like Asterisk to be able to
present a call to the Hicom, either
2009 Feb 25
4
switchtype QSIG and Asterisk implementation
Hi,
Is Asterisk "fully QSIG-compliant"?
I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4.
Zaptel versions are 1.2.26 and 1.4.11.
I am using switchtype=euroisdn and all works fine.
However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by
2013 Oct 31
2
issue with dahdi_channels.conf
Hello list
i have an issue with my dahdi_channels.conf
in span 1 when i use it like below i can do my outband calls without issue
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 17-31
context = default
group = 63
but when i add in channel 1-15 like: channel => 1-15,17-31
i receive all
2006 Jan 12
1
PRI and QSIG
Hi all,
I'm planning on connecting our Asterisk to our legacy PBX (an Avaya INDeX).
I was originally going to sit it between our ISDN connection and the INDeX
(tried it, worked ok) but now I intend to hang it off a spare PRI card just
so should the * fail we keep our ISDN's at the INDeX, yes I've looked at
ISDNguard but little info from the manufacturers.
Anyhow, I'm now
2006 Jun 23
3
Asterisk-1.2.9.1 with Siemens HiPath 4000
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one
upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000
in ISDN PRI with protocol QSIG, the one that is happening he is that the
calls originated for PABX Siemens and destined to SIP phones asterisk are
being without audio, nor Ring, is dumb. They could help in this case me?
Best Regards
Josu?
2006 May 03
4
QSIG support in Asterisk
I am looking to get the info about QSIG support in Asterisk.
Does Asterisk have QSIG support?
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?
If so, How to configure that?
Thanks
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2006 Jun 26
2
Asterisk x Siemens HiPath 4000
Hello all. I have installed and functioning asterisk-1.2.9.1 where I
effected one upgrade in asterisk-1.0.9, is interconnected with a PABX
Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is
happening he is that the calls originated for PABX Siemens and
destined to SIP phones asterisk are being without audio, nor Ring, is dumb.
They could help in this case me?
Best Regards
Josu?
2006 Apr 27
5
PRI configuration
Hi,
I am getting this message on the * console on my first pri span. Pri
show span show it is down, and I can't make any calls from the span.
Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Apr 27
2006 Oct 11
0
Hicom 150 -- BRI -- Asterisk
Hi,
Is is possible to implement this:
Hicom150 --- BRI (QSIG) ---- Asterisk
I've been reading Siemens documentation and they say:
"Digital nailed connections
Corporate communication networks can be implemented over digital S0 or
S2M nailed connections between several Hicom systems using the CorNet N
protocol and between Hicom and non-Siemens systems using the QSig protocol.
The
2006 May 27
2
amportal doesn't start with brestuff(ISDN)HFC-PCI
Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design
GmbH ISDN network
controller [HFC-PCI](rev 0.2)
This is how I installed bristuff:
how to install hfc card
after unload asterisk and amportal whit
amportal stop
type "setup"
unselect zaptel in system service...
and set the lan
--->reboot<---
cd /usr/src
wget
2005 Mar 19
2
Goto and E1 line
Hi,
I have a server with 2 TE110P cards. 1 card is plugged to telco line,
another card is plugged with a Hicom PBX.
I want to send some call to VoIP phones and all other to my PBX.
I don't known how to make my dialplan :
===========Extensions.conf==========
[incoming_call]
exten => 090200000,1,Goto(callcenter,100,1)
exten => 022956353,1,Goto(callcenter,100,1)
exten =>
2009 Oct 05
1
Peculiar error message when using Q-SIG
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! < Unknown IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932 specification, so I don't understand why
this error is occuring.
2013 Apr 30
2
Asterisk QSIG doesnt send the calling name to Nortel CS1000
Hello to all,
I have a problem with an asterisk qsig.
I have three machines:
Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk--->
Asterisk
I use Snom phones on Asterisk.
If I call from Asterisk to Nortel, Nortel reminds me of the name of the person
i'm calling and I visualize on the display of Snom phone, but if I call from
Nortel to Asterisk, the QSIG does not send
2004 May 09
1
No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier ----S2M------ * -----S2M------Siemens
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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 Feb 10
1
QSIG error -- can somebody explain?
Hi all,
I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX
via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the
outside world and should forward our calls to the telco. This setup
works correctly as far as I use euroisdn as the switchtype.
The first problem was that it is only possible to run the * side in
CPE-mode -- I wanted NET.
Anyway, I configured * this way:
2008 Nov 06
1
ISDN Cause Code 100, Bosch Integral Management Connection
Hi all,
first off all - sorry for the cross posting - i did already posted this
message to asterisk-dev - after that i realized that it isn't really a
-dev related question - more a -users questions. So ignore it on -dev ....
we have the following setup
PSTN 3 PRI Lines <---> Asterisk (1.4.22) <---> Siemens HiCom
<---> Bosch Integral
The Asterisk Machine
2005 Jul 07
1
experience with analog channel banks in E1 land
hi,
we are currently planning are large site which will migrate from an old
siemens hicom pbx to asterisk.
it will be a slow migration, the asterisk server will be inserted
between the telco E1 and the hicom. new phones will be sip ones.
the customer has several fax machines and analog phones (some of them
have to be explosion-proof). around 50 analog ports in total are needed.
as we are in