Displaying 20 results from an estimated 4000 matches similar to: "Call recording"
2003 Mar 20
11
Voicetronix
Has anyone gotten the voicetronix boards to work with Asterisk, what
would it take? Or does anyone know where I can get 4 ports or more fxs
PCI cards that do work with asterisk?
Brian J. Schrock
Network Engineer, RHCE, CCNA
Anistone Technologies
Phone: 614-798-9106
FAX: 614-573-7165
6926 Avery Rd.
Dublin, OH 43017
2003 Apr 02
7
FAX over IAX
Hi,
We are looking at consolidating our lines with PRI. This will allow the
elimination of many fax lines. Some of them will be replaced with this type
of config ...
PRI * IAX * Channel-Bank FAX
We will have daggressor suppressor enabled. Is anyone doing this and should
I expect smooth operation?
John
This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.
2004 Sep 02
2
Sorry, Newbie here
I never heard of Asterisk before today, but from what i'm looking at on the website and hearing, it sounds pretty incredibly. If I understand correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux Server, and a couple IP phones or Netmeeting on a few workstations, and of course, Asterisk which is free; I call have a small call center.
This can't be? I was looking at
2003 Apr 16
1
New TDM400P no dialtone
Hello,
Does anyone know what may be causing this? Asterisk was built from cvs
tonight. Ztcfg also says us is an invalid tone zone. Anyone got some
information on what this is, why is it happening, and possibly some
solutions?
[root@anistonetech zapata]# asterisk -f -d
DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing
/etc/asterisk/asterisk.conf
DEBUG[1024]: File config.c, Line 653
2003 Oct 28
4
Software FAX
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.....
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include files to
/usr/local/include
2) build/install spandsp
3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree.
4)
2003 Mar 07
3
ISPs with QoS for VoIP?
I'm wondering if anyone knows of ISPs with service that has QoS
features that would be good to use with VoIP stuff. Granted,
the QoS would only be supported as long as you stayed within
their network, but it might be better than nothing.
--Eric
2003 Oct 06
1
SIP X100P Echo Problems
Like most others on this list I also have some really annoying echo whenever
a call goes out to the PSTN from a SIP phone...
SNOM/Budgettone -> Asterisk -> X100P -> PSTN
I have tried every echo canceler in the makefile and turned on and off
aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and
I can get it reduced to only a few seconds on the intro of the call and
2003 Mar 06
2
Dial Problem
I have a simple problem with sialing a SIP device. I'm SURE
it's a syntax problem, but I dunno what it might be.
Here are the debug messages:
== Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>)
-- Executing Goto("Zap/1-1", "2111|1") in new stack
-- Goto (default,2111,1)
-- Executing Dial("Zap/1-1",
2004 Sep 14
4
Sending Caller ID info in MD/USA
All,
Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's
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
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 Sep 24
3
list of voice prompts
Does there exist a text file with all the 'standard' Asterisk voice
messages? I'm planning to get them recorded in dutch, but need to know the
exact text of each prompt...
Michiel
2003 Mar 05
1
Sip registration Timers
Hello,
I have my sip stuff seemingly working fine as well as my zaptel stuff
working great... But I have a problem with sip registration timers (I'm
guessing here).
In my extensions.conf file I have this...
exten => 2244,1,Dial,Zap/2|25
exten => 2244,2,Dial,Sip/brian|25
exten => 2244,3,VoiceMail,u2244
But if I close my sip phone and a call goes through it will still wait
the 25
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
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 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 Oct 14
6
WCFXO echo rexolved for me
Hello,
I resolved my echo issue using grandstream/estara etc etc sip phones and
wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl
motherboard for an asus p4pe? i845? based motherboard and the echo has
completly gone away along with aggressive suppressor option in the makefile.
I hope this helps others.
Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH