Displaying 20 results from an estimated 6000 matches similar to: "Problem on Zap Channel with IVR"
2007 Sep 21
0
Problems bringing up ZAP trunks via PRI
Hello,
I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based
email to fax gateway. At this time, I have a ZAP PRI link between the
eFax server and my VoIPSwitch. The ZAP channels are configured, the B
and D channels are up, and I have green link lights on either end of
my cabling, but when I dial the number I have assigned to my eFax
server, the call never seems to route
2005 Jun 07
2
Help! Zap echo on bridged calls
I've been going nuts lately trying to get rid of an annoying echo
problem that makes my asterisk server unusable when clients try to call me.
Here's the breakdown of the issue - Hoping that someone can throw me a
clue:
My setup is as such:
Single AMD Athon machine with X100P clone card and voip through multiple
providers .
* Inbound calls through the X100P that do not bridge to
2007 Jun 26
0
No CID on Zaps - TDM400
I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs).
With Trixbox out of the mix and a regular phone connected I get the CID
fine yet Trixbox shows 'unknown':
dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is 'ringall'
Here is my Zapata.conf if it helps:
#############################
;
; Zapata telephony
2007 Jul 24
1
[beginner] Problem of detecting call
Hello,
I have some problem to start asterisk.
First I have followed a lot of tutorials to complete correctly the install
process. Now it works when I type zttool I can see when I am or not
connected to the PSTN.
But, I run asterisk with vvvv verbose and I can't see the call detection.
There is no detection of the call.
I have a X100P card FXO with only one line. So only one channel
I
2007 Apr 04
0
Bad Line Noise over T1
I've got a system where I'm integrating a Nortel Option 11c with a
Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell
PowerEdge 350)
We've got things mostly up and running and all seems well... except...
If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where
xxxx is an extension on the Opt 11, the call goes through to the Opt 11
but I have terrible
2008 Nov 12
4
The sound is played but I did not hear
Hello,
I have another little problem with my ZAPs channels, in fact, when I
received a call, I heard no sound while in the CLI, sound is played:
-- Starting simple switch on 'Zap/4-1'
-- Executing [s at from-zaptel:1] Answer("Zap/4-1", "") in new stack
-- Executing [s at from-zaptel:2] BackGround("Zap/4-1", "hello-world") in new
stack
--
2005 Mar 26
0
Echo on Zaptel hardware (Wildcard 100XP)
I've enclosed by config... I've tried everything from lowering the
tx/rx gains.. to toying with 32/64/128 echo canceling taps... at 256
echoing is really bad...
I've even tried recompiling the zaptel driver with the MARK2 super
echo canceling support...
I still have a very slight echo that I simply can not get rid of....
this echo is ONLY when going VoIP to POTS.. and I understand
2005 Jun 20
0
Can't get TDM04B to work!
Can't get a Digium TDM04B working. Asterisk is running. I seem to have setup the trunks OK. But whenever I make an outgoing call get the 'all circuits are busy now' message. If I call in nothing happens at all!
Here is my zapata.conf file:
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
fxsks=1-4
2006 Mar 17
0
caller unable to transfer
Hey all, posted this the other day, but re-read it & realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using tT but transfers are only available when I'm the recipient of the call, not
2006 Apr 04
0
some problems with asterisk and E1
Hi,
I am using asterisk 1.2.5 and have some problems with asterisk connected with
an E1 card to our PRI. Dialling in and out generally works. When someone dials
in from a mobile phone, all numbers are sent as a block, and the called
extension rings as intended. when someone picks up his phone handset, waits
for a dial tone, and then dials in manually, the call will be redirected to
the
2007 May 23
0
IVR Loop on invalid input
We are running 1.2.14 with an IVR in the dialplan.
If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a
couple of digits very rapidly (I found this with 33 on a sticky keypad)
which are an invalid response, Allison will go into a loop saying 'I'm
sorry, that is an invalid response, please try again.' over and over.
This does not happen with a commercial
2005 Sep 28
1
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'
Any ideas?
51] logger.c: [chan_zap.so] => (Zapata Telephony)
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing
2006 May 01
1
unable to set outgoing callerid
Hi *,
now for a long time i am trying to set the outgoing callerid, without luck.
I am here in Germany, my asterisk has a pri interface connected to a PMX
installed by Telekom. All telephone calls are preselected to EcoVoice.
I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2.
A week ago we tried with a device able to simulate a telephone system so send
out a callerid, and that
2007 Aug 09
2
Terrible clicking on T1
Hey All,
I have an Asterisk box connected to a Nortel Option 11C via a T1. In the
Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI
card. The Nortel is also hooked to the PSTN via a T1 on a different
NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files
below.
Our issue is that when a call is sent over the tie line between the two
systems, the audio on the
2006 Mar 24
0
UK pri almost working
Hi all I wonder if anyone can give a little insight into this;
asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless,
ISDN30. 10 zactive channels
Incoming calls work fine no problems tested 43 DIDs all working.
zaptel.conf
# Global data
loadzone = uk
defaultzone = uk
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
dchan=16
zapata.conf
[channels]
language=en
context=from-pstn
2005 Sep 28
2
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'
Did you compile and install libpri *before* Asterisk? I had same problem
(among others) b/c I didn't install in the correct order. Try the awesome
asterisk_update.sh shell script.
Are you trying to emulate CPE or NET? Try signalling=pri_cpe
Check for whitespace behind the statement, zapata.conf seems bitchy about
whitespace.
hth
-----Original Message-----
From: Steve Totaro
2006 Apr 13
1
Ztmonitor shows RX is always on.
Details:
Asterisk 1.0.9
Zaptel 1.0
Dell P3 1ghz with X100P Clone
Location: India
This is an interesting issue where when I open up ZTMonitor, it shows the RX
as being on. It seems that Zaptel doesn't know to hang up the line so after
a couple of hours when the telecom cuts the line, everythign stops working.
Things I've tried include playing with the zaptel.conf, trying zaptel
v1.2(with
2010 Aug 23
1
Dahdi install gone wrong
The card you installed has FXO or FXS modules in it ????? are you getting
your lines directly from the telco co???
Doug D
On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent:
* -----Original Message-----
* From: Todd Reese
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion
* To: asterisk-users at lists.digium.com [3]
* Subject: [asterisk-users] Dahdi
2005 May 20
1
MFC&R2 Venezuela with libunicall
Hi,
I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code.
All libs compiled successfully and the E1 have a green light!
I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working.
My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P
2005 May 31
0
Re: Asterisk-Users Digest, Vol 10, Issue 234
Hello All
I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error.
error messages:
*CLI> Warning, flexibel rate not heavily tested!
Rx CAS bits 0x9 [ 10000/ 0/ 0]
Line unblocked
-- R2 Channel 4 unblocked
Rx CAS bits 0x9 [ 10000/ 0/ 0]
Line unblocked
-- R2