Displaying 20 results from an estimated 1000 matches similar to: "isdn cli"
2004 Jun 23
6
Outgoing CLI
Hello
I have contacted my line provider who is saying that in order to get my 0845
or 0870 number to id as the incoming number on a landline that i may call i
need the following.
User must provide - NPI set to E.163/E.164
User must provide - TON = "national or international
I have had a good search around and can't seem to find a good answer to
this. Does anyone have any idea where i
2010 Feb 17
4
Unrecognized prilocaldialplan NPI modifier
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
prilocaldialplan NPI modifier: k
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
prilocaldialplan NPI modifier: o
[Feb 17
2003 Dec 20
3
ivr key press?
I'm testing an ivr implementation (first time) using:
exten => 620,1,Wait,1
exten => 620,2,Answer
exten => 620,3,DigitTimeout,5
exten => 620,4,ResponseTimeout,10
exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for"
exten => 1,1,Goto(npi-directory,s,1)
For initial testing, I've arbitrarily mapped this onto ext 620 (will
change that later
2019 Jan 15
2
Cannot access other computers on LAN
Hello Julien,
Am Mon, 14 Jan 2019 22:15:47 +0100
schrieb Julien dupont <marcelvierzon at gmail.com>:
> ** Test 1 **
> On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3'
> When pinging 192.168.1.1 from client 1, with no success, I see no packet
> passing.
Sorry - the tcpdump command should end with "192.168.1.1" instead of
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.
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)
2018 Jan 15
2
Digium G100 and CID Dropping First Digit.
Hi All,
I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was previously using DAHDI and a TE121 device which may have been adding the 0, I'm not too sure about
2005 Aug 12
8
Incompatible destination (88) Error Message
I have connected asterisk 1.0.7 with Avaya Definity via E1 with a
TE100P Digium Card.
Inbound calls are working perfectly and I dont have any problem. But
when I try to make an outgoing call with my softphone (xlite) I am
getting the following messages.
Hungup 'Zap/13-1'
Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack
Called g1/3118
Channel 0/1, span 1 got
2007 Jan 18
1
Passing video calls / bearer capability thru PRI
Hi all,
using latest asterisk-svn
I want to reflect an video call incoming via an PRI EuroISDN channel to
another outgoing PRI channel,
and I want the the outgoing channel to have the exact same bearer
capability
< Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
< Ext: 1 Trans mode/rate:
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
--
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
2004 May 10
1
Callerid via PRI
When receiving multiple calling numbers via a PRI for a call setup, I
cannot find the ability to select between either first or last. Is
there a way to do this currently? If not, can anyone help me in getting
this to work? Here is the dump of the PRI in intensive debugging.
Please note that I want the first calling number, not the last. Any
help would be much appreciated. Below the PRI trace
2006 Apr 19
2
PRI caller ID
Below is a snipped debug on our PRI. We are getting number only for the
CallerID but the telco says they are sending us Name and Number. We are
getting the Name in a second frame but Asterisk is not passing it to the
device it rings. The message below says "Presenation allowed of network
provided number" which leads me to believe Asterisk thinks it should not
be displaying it. Can anyone
2006 Apr 29
2
problame with outbound calls on pri
Hi. recently I have been trying to setup a PRI on asterisk. Inbound
calls are working just fine but I am not able to make outbound
calls. Does anyone know what I need to change to make outbound
calls work? Right now the PRI is instantly hanging up on the outbound calls.
I have included full debug info as well as config files.
/etc/zaptel.conf
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
2005 Sep 28
1
Asterisk does not send "Setup acknowledge" on euroISDN E1
Hello,
Configuration:
Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1
I am trying to sort out the problem:
1. Provider's switch sends "SETUP";
2. Asterisk receives "SETUP", rings allocated extension but does not
send "Setup acknowledge" (or any other messages) to switch;
3. After 4 seconds of waiting of *'s response switch sends
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
2007 Feb 28
2
No Caller ID Name PRI NI2
I there,
I have some trouble to do working caller id name for outgoing calls on
the PRI we just installed. Caller id name work on incoming calls only.
Caller id number work on incoming and outgoing calls.
The provider, Goup Telecom, said that's in what i'm sending. They said
that I send the cid name in ascii code and to do it working, I need to
send it in hex.
So I take some traces
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
2019 Jan 14
2
Cannot access other computers on LAN
Hi Julien,
Am Mon, 14 Jan 2019 18:04:40 +0100
schrieb Julien dupont <marcelvierzon at gmail.com>:
> Investigating with tcpdump withoug guidelines is beyond my skills I'm
> afraid.
Try this on your VPN_office host:
tcpdump -npi any icmp and host 192.168.1.3
In parallel you start a ping from the other network:
ping 192.168.1.1
I assume, that tcpdump will show all packets from
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