Displaying 20 results from an estimated 4000 matches similar to: "ZAP channel on TE410P doesn't hang up"
2005 Feb 16
0
ZAP channel on TE410P doesn't hang up (Plain Text this time)
Hello * users
Sorry I forgot to send the mail in plain text the first time...
I've have a rather disturbing problem, which I don't know how to debug or
how to solve, but first a brief description of the set up.
One Asterisk server with a TE410P card installed (first line used on this
only), and a number of Wellgate 3504A (4 port FXS devices with SIP
firmware). There is no connection
2003 Sep 03
1
MusicOnHold and MP3Player not triggering "answer"
Hi
I have kind of an odd problem.
When dialing in from an outside line via a TE410P card it seems like
MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote
end which is calling * doesn't hear the music but just keeps ringing. But if
I insert a Playback("file_which_dont_exist") just before the Moh or
MP3Player I can hear the music. Actually I observed the
2003 Sep 03
0
Problem with Asterisk -> Welltech Wellgate 3502
Hi there
I have some problems in setting up a connection between * and a Wellgate
1501 (1FXS). Either I can dial from * to my Wellgate or I can dial from the
Wellgate to *.
On the Wellgate you have to setup both "Line number" (which is the number it
answers) and "Line account" (which is the account it registers as). Should I
expect any problems if these are set the same?
2003 Jul 02
0
Asteriks, GnuGk and outgoing calls
Hello there
I'm quite a newbie in the IP Telephony area. I'm playing a little around
with a setup with one linux box with a e100 p card installed, which
functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper).
I have two h323 phones, Welltech WellGate 1501 and 3502.
So far I've managed to get the two IP phones and Asterisk connected to the
GK. I can place calls from one
2003 Oct 16
2
Problems with TE410P and E1 line --> Unable to open D-channel 24 (No such device or address)
Hi everybody
I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri.
Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message:
-------------------------------------------------------
-- Registered channel 1, PRI Signalling signalling
.....
-- Registered channel 15, PRI Signalling
2003 Nov 04
1
Flash hook -> SIP device
Hi there
I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device.
I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this.
What is happening when you flash hook, I
2003 Sep 04
1
SIP - DTMF Payload type
I have a problem with my Welltech Wellgates.
I can't call any extension which starts with or includes * or #.
When dialing it responds fine but after some seconds I just get a busy tone
and on the Asterisk console it says "SIP/2.0 484 Address Incomplete".
Don't know if it connects to the DTMF payload type.
Yesterday I made som different tests and observed that if DTMF payload
2005 May 30
0
IAX2 to H323
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected.
When I dial from SJPhone (H323) ->
2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
Hello all
I've discovered that SIP channels sometimes get stuck in *.
I've read some posts from Fri 29 Aug 2003 which mentions this issue, but
there doesn't seem to be any final answers
I don't know if this is related to the 0001604 bug?
Below is a list from one of the incidents:
I know the (d) means that it is scheduled for destruction but the 10.1.1.45
channel hasn't
2003 Oct 29
1
Distinguish between voice and data call
Hi
I have an Asterisk installation with some SIP and MGPC devices, and I also have a TE410P on a E1 line.
If I make an outside ISDN data call to asterisk the phone rings as usual and if I answer it, I just hear some clicks.
I've read that the D-Channel has information about the call, if its voice or data.
Is it somehow possible to end/ignore this call already before it is ringing?
2003 Sep 01
6
Change include contexts runtime
Hi there
How do I change the dialplan runtime, if I for example wants all calls on
the main number to be answered by a voicemail (when it is out-of-office
hours).
I want to be able to change the configuration by pressing a DTMF combination
e.g. *82. Can't figure out whether it is necessary to change contexts or how
to do it.
I have read a lot of examples and config documentation, but I
2003 Oct 22
0
Different MGCP issues
Hi there
I've installed a 12 port MGCP gateway, (Hitron MDU-5612), which works ok most of the time. Sometimes when talking to the outside world, (via a TE410P), the line gets disconnected. I think its related to MGCP because I've also setup some SIP devices which doesn't behave like this.
I've examined the logs but can't find anything useful, it looks like its the MGCP device
2006 Apr 11
2
Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4
ports, 31 channels each and able to receive incoming calls and fax
perfectly.
I've done this in my dial plan.
exten => 111,1,Answer()
exten => 111,n,Ringing()
exten => 111,n,Wait(2)
exten => 111,n,AbsoluteTimeout(30)
exten =>
2006 Apr 11
1
AW: Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany)
marcus
-----Urspr?ngliche Nachricht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
Gesendet: Dienstag, 11. April 2006 16:33
An:
2004 Sep 08
4
WellGate 3504A with Asterisk SIP authentication and config
hey * folk,
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
[1235]
host = dynamic
secret = somepass
context = default
type
2004 Oct 06
2
Working Wellgate *SIP* 38xx/35xx hardware anyone?
I'm loosing hair at cosmic speed now for the past 10 days.
Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have
very buggy firmware possibly due to hastely done porting from H.323
firmware.
Is there anyone on this mailing list who was able to:
1. setup a 35xxA FXS with all ports authenticating properly with *?
or
2. setup a 38xx FXO to work as dial-in from pstn to
2006 Apr 07
1
wellgate registration 3802
I have a new wellgate 3802 unit. I have not gotten it to
register with asterisk 1.2.6.
My proxy setting is the correct IP in the 3802.
My security config is 1001/1001 and 1002/1002 on the wellgate (simple at
this time).
My sip.conf has:
[wellgate3802L1]
type=friend
dtmfmode=inband
username=1001
secret=1001
host=dynamic
canreinvite=yes
nat=no
context=wellgate
[wellgate3802L2]
type=friend
2008 Nov 27
2
Wellgate & Asterisk
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
1. Sip Config
Mode: Proxy
Primary Proxy IP Address: *.131
Primary Proxy port: 5060
Line1 Number: 1002
2. Security Config
Line1 Account: 1002
Line1 Password: ******
3. Line Configuration
Line1: Type=FXO, Hunting Group=2, Hot Line =
2005 Mar 18
2
Pattern matching in extensions.conf
Hello fellow * users
Hope this isn't a stupid question; I've done my research but could not find
a proper answer.
I have 8 different destinations which I want to match. The numbers are:
###### 00
###### 20
###### 30
###### 40
###### 15
###### 35
###### 12
###### 44
Right now I've solved it by doing this:
exten => _######[0234]0,1,HangUp
exten => _######[13]5,1,HangUp
exten
2003 Sep 04
1
I don't think I understand "Call pickup"
I must be getting something wrong about this call pickup.
In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I
call from my mobile to * and then try to dial *8 from any other phone than
the one which is ringing I just get a "Nothing to pick up" answer on my *
console.
I also have experimented with those parameters in sip.conf but are not aware
of exactly where to