Displaying 20 results from an estimated 200 matches similar to: "list of voice prompts"
2003 Sep 22
3
iaxtel and iax.conf
I have tried for over a month off and on to get iaxtel for inbound to
work... and tonight after alot of troubleshooting we noticed this:
iaxtel inbound will use the last entry in your iax.conf to auth against.
So if [iaxtel] is at the top and say [voicepulse] at the bottom. An
inbound call will try to auth against that [voicepulse] entry even with
the [iaxtel] entry at the top of the file. Has
2003 Oct 08
1
BudgeTone 102 flakey sound
I have experienced lots of apparently dropped packets (in other words,
lots of short interruptions of what the other party tries to tell me)
with a GS102 and chan_capi. The GS102 is connected through a lightly-loaded
switch directly connected to the * server, so bandwidth/latency
shouldn't pose a problem. Funny thing is that the switch indicates
10mbit on the GS102 port - is that correct?
2003 Nov 25
10
PCI 3.3 V
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
Cristian VASILIU
AccessNET International S.A.
Software Programmer
mail to :<cvasiliu@accessnet.ro>
www:<http://cvasiliu.home.ro>
2003 Oct 20
26
Survey: Grandstream improvements.........
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and such.
So please rate your ideas on a scale of 1-10
1 = Nice to have some day
10 = Got to have it right now
Things
2004 May 17
2
Problems w. chan_capi + ztdummy
Hi Everybody
I've got a weird problem. I am running one Asterisk system on a dual
processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN
card installed with latest drivers. Dialing out through the ISDN cards from
an internal Snom phone works fine and so does dialing in. Except - if I
load the ztdummy module (for IAX channels) the capi drivers starts acting
up. It is hard
2003 Jul 11
4
module : cdr_sybase.so
If anyone is interested ... just in case! :-)... I have tried to write ,
based on the cdr_mysql.so module, an Sybase module.
To compile you can use something like that:
export SYBPLATFORM=linux
export SYBASE=/opt/sybase
cc -I$SYBASE/include -c -o cdr_sybase.o cdr_sybase.c
cc -shared -Xlinker -x -o cdr_sybase.so cdr_sybase.o -lsybdb -lm
-L$SYBASE/lib
(anyone could write the corect Makefile
2003 Jul 11
7
ISDN PRI E1 configuration with E100P
<P>hi Everyone,</P>
<P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P>
<P>*CLI> <BR> == D-Channel on span 1 up<BR> -- B-channel 1 successfully restarted on span 1<BR> --
2003 Jul 15
1
Alphanumerical digits
Sorry Martin to bother you again!
I have an ISDN flux with 100 numbers. The local PSTN is sending now the
DNIS/DID (so they said!!!) (I have set the immediate=no in zapata.conf)
but I have the same problem as before :
NOTE : the number is alphanumeric-DID alphanumeric (I will make tests
with numeric mumber!).
WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified,
but not
2003 Sep 23
1
Question about dialogic hardware
1. D/120JCT-LS card with 12 ports. This ports are FXS ports?
2. It is true that "Dialogic drivers cost of $15 per channel" ?
3. Can I use this hardware with asterisk (for E1-ISDN using Wildcard
E400P <http://www.digium.com/index.php?menu=wildcard_e400p>) ?
4. Anyone with experiance can tell me how they work and can provide a
configuration example? (2 DSP Motorola procesors - I
2003 May 18
1
DECT to Voip gateway
This looks like a fun box... a Voip to Dect gateway, I've mailed them for
pricing details....
< <http://www.computex.com.tw/news_archive_detail.asp?index=4053>
http://www.computex.com.tw/news_archive_detail.asp?index=4053>
Betel Consultancy
Abelenlaan 19 T: +31 20 640 3018
1185 RT Amstelveen E: <mailto:info@betel.nl> info@betel.nl
The Netherlands W:
2003 Aug 11
1
zaptel sync
Simple Q but I can't find the answer in the archives (and am too lazy to
look in the source, but then its 32 Celcius here...
Do all digium cards provide the zapata timing? e.g. also the analogs
(including the X100P) or only the E1/T1 -ones or do I need to use ztdummy on
the analog cards?
Thanks,
Michiel
Betel Consultancy
Abelenlaan 19 T: +31 20 640 3018
1185 RT Amstelveen
2003 Aug 25
1
chan_zap.c zt_rec: Unknown error 500
Hi all,
I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a
channel bank and a digium E1 connected to the PSTN.
I get occasional warnings from asterisk:
WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error
500
This happens mosttimes in a loop like this:
[netland_helpdesk]
exten =>
2003 Jun 26
1
Retry dial when busy
Some switches provide the functionality to try a number till it becomes
available. Thus when one dials a number and get a busy, one enters a *XX#
code and the switch will call your extension when the called party becomes
available. Has somebody already built this in/for Asterisk, otherwise I'll
look into it.
Michiel
Betel Consultancy
Abelenlaan 19 T: +31 20 640 3018
1185 RT
2004 Sep 23
1
PRI(E1) Call recording with Digium cards?
Hi,
I've been asked to see whether it is possible to do call logging for call
center environments at a lower budget than the usual $1000 per channel.
Afaik, with PRI this is possible through a high-impendance Y connection,
but I wonder whether this would work with the Zapata cards. Anyone ever
tried this?
Regards,
Cees
--
XP SP2 can cause cancer in rats
2003 Jun 12
1
srv.c + srv.h
I just downloaded the latetst CVS. A compile now complains about a missing
srv.c & srv.h used in chan_sip.c. Can they be added?
--
Betel Consultancy
Abelenlaan 19
1185 RT Amstelveen
The Netherlands
http://www.betel.nl
tel. +31 621 858 469
2003 Mar 05
17
Call recording
Hello,
How would I go ahead a record all phone calls into and out of my
asterisk server. I know the legality issues behind it, but I could
always play a recording to let people know they will be recorded.
Brian J. Schrock
Network Engineer, RHCE, CCNA
Anistone Technologies
Phone: 614-537-2817
FAX: 614-573-7165
6926 Avery Rd.
Dublin, OH 43017
2003 Aug 31
2
DBSaveTree & DBLoadTree
Hi all,
Has anyone already written something which allows saving and loading the
internal DB settings? All users CFWD and speeldial settings are stored in
the DB in my setup which makes it a pain to restart Asterisk....
Looking at showtree in db.c (why isn't that exposed in the CLI?) It
shouldn't be too difficult, but I don't want to reinvent the wheel.
On the same track, I am also
2003 Dec 15
4
transfer with threeway calling
Hi,
We are using threewaycalling & flash transfers over a CAC channelbank.
The following happens:
Call comes in to my extension
I talk to a party and press flash
party goes on hold, I get get dail tone
I dial internal number
internal party answers
I press flash once more
we are now in a three party conference
Or I hang up, and thus transfer the call.
Thats fine, but....
What if the
2003 Mar 13
1
E1 yellow alarms
About every hour I see the yellow alarms on all or a number of channels of
my PRI which is connected to the dutch telephony network, Asterisk keeps
on working fine....
Here's an example where channel 1-24 went into alarm:
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
alarm on channel 1: Yellow Alarm
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):
2003 Nov 25
2
zt_rec: Unknown error 500
I have a number of Zap/ extensions defined in a queue with ringall
strategy. When this queue is called sometimes Asterisk seems to think
that one of these channels is busy, while it is NOT. The following is
shown on the console:
--Called 44
-- Called 36
-- Called 41
-- Called 35
-- Called 38
-- Zap/44-1 is ringing
-- Zap/36-1 is ringing
-- Zap/41-1 is ringing