We have recorded wav files with 44k, 22k, 16k, 11k and 8k
Asterisk does not accept these wav files. I used sox input.wav
output.gsm to get them to work.
However, the only the 8k file did convert and the quality is poor. How
can I improve the quality?
bye
Ronald
I know I can setup asterisk without Internet at all and it works as
local pbx.
Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?
bye
Ronald
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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:
Guillermo Salas M. wrote:
El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió:
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):
I've one wellgate 3804 (old version
Ronald Wiplinger (Lists) wrote:
During compiling I have not seen an error, however, when I start
asterisk again it ends with:
app_morsecode.so = (Morse code)
== Registered custom function 'SYSINFO'
func_sysinfo.so = (System information related functions)
Segmentation fault (core dumped
During compiling I have not seen an error, however, when I start
asterisk again it ends with:
app_morsecode.so = (Morse code)
== Registered custom function 'SYSINFO'
func_sysinfo.so = (System information related functions)
Segmentation fault (core dumped)
How can I figure out what is wrong?
we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line
we have for our office a different ADSL with one IP shared.
Two identical setup snom 360 (except the user name) with two public IP
addresses are connected at the hub to the server / DSL line
phone A can call B, B cannot
I used to run an Asterisk server in the office, ... was looking for a
small replacement. I am not sure if that one is a good idea yet either.
How about this one:
I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is
an Asterisk server.
Is there a Asterisk hoster out there?
I had installed in the office an Asterisk server, but the company is
gone and I could keep the server.
However, for my family with three members and two phone lines this
server is overkill. I am looking for a compact solution, which is more
suitable for me.
I want a small silent box, which can
I have a local asterisk 1.2 and a remote asterisk 1.4.
Snom 190 can be used with the local asterisk but not with the remote one.
I need some hints where to track down this issue.
Some information:
Snom 190:
Line 1:
Account: 615
Password: OnlyIknowit
Registrar: ast.mydomain.com
, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTNAME=Ronald Wiplinger) in new stack
[Jan 27 16:03:32] -- Executing NoOp(SIP/88621001-00728610, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTCOMPANY=Elmit.com) in new stack
[Jan 27
I have a AMD64 CPU and use SuSE 9.2 with kernel 2.6.8-18
I tried to upgrade svn version 1.4.x but it fails at each part and
mainly because the system is with 1100 days getting to old.
I have to make a decision and need your advice.
CPU AMD64 3200+
1 GB RAM
Digium card with 2 FXS and 2 FXO
Ronald Wiplinger wrote:
Can anybody give me a hint, please.
I have a Welltech FXO device and from PSTN coming calls will be
transfered to the extension number 1001.
I want that the caller can reach the extension number by dialing
said number.
My 1st try was:
exten = 1001,1,NoOp
I have this in my extension.conf:
[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Wait(1)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include = fax2emailstart
[fax2emailstart]
exten =
I need to setup multiple fax extension numbers.
What is the best way to do that?
It should send the fax as pdf to the assigned email address (or
addresses) of that extension number.
It should also move the fax to a web site for online view.
It should - if possible - try to make OCR text file as
1. Instead of using *1 (automon) I need to record each phone call at a
certain * box.
2. While already talking about this. I want to autodelete with cron at 2
am in the morning all recordings which are older than 50 hours! How can
I do that?
bye
Ronald
I haven't updated for a while and when I looked on the web site how to
do a SVN update, I cannot find it anymore.
CLI show version
Asterisk SVN-branch-1.2-r42600M built by root @ asterisk on a x86_64
running Linux on 2006-09-10 22:52:42 UTC
1. Where is the description for the SVN update now?
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
How can I set-up a MRTG with 4 graphs, whereby:
1 data in
2 data out
3 ONLY voice(/video) data in
4 ONLY voice(/video) data out
bye
Ronald Wiplinger
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Can anybody tell me how I can set-up multiple access points with
overlapping coverage, so that a moving WiFi phone user can continuesly
use the phone.
bye
Ronald Wiplinger
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Where can I get a starting point for setting up sms via VoIP and via web.
I want to send SMS from VoIP or web to VoIP phones and GSM phones.
1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones can be used?
bye
Ronald
bails wrote:
Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There
is a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of monkeys recording.
The best part
it records the conversion.
What do I miss?
bye
Ronald Wiplinger
On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
and where should Set(DYNAMIC_FEATURES=hangup#play#testfeature)
be
and I want that only 601 and 621 can use this feature.
bye
Ronald Wiplinger
The best part of it is that you can hang up and the recording will
continue to play to the caller. When it expires, so does the call
On 11/11
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
bye
Ronald Wiplinger
On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I want to add some sound filed on demand
each!!!
Either the caller-id is not set, is 0 or is a tollfree number.
bye
Ronald Wiplinger
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Brian Capouch wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is it exclusive? Either Realtime or priority n ???
If so, what is the better way?
I believe 'n' is just a shorthand way of writing previous line + 1,
and gets converted
Is it exclusive? Either Realtime or priority n ???
If so, what is the better way?
bye
Ronald
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I want to make the context [default] as an alarm, for not having
set-up correct.
I am looking for a way to get incoming calls via ENUM or via names (e.g.
sip:[EMAIL PROTECTED]) into a defined context. How can I do that?
bye
Ronald
___
How do I use priority n correct?
Here is the current example:
exten = 615,1,Dial(${PHONE_615},60,tr)
exten = 615,2,Voicemail,[EMAIL PROTECTED]
exten = 615,103,Voicemail,[EMAIL PROTECTED]
and:
exten = 617,109,GotoIf($[${DIALSTATUS} :
(CHANUNAVAIL|CONGESTION)]?110:999)
exten = 617,110, .
When I reloaded my asterisk I saw these lines, which I have noticed before:
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039
add_realm_authentication: Format for authentication entry is
user[:[EMAIL PROTECTED] at line 797
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039
I am thinking if re-invite will interfere accounting.
Please help me to figure it out:
Phone A is registered at asterisk and calls a gateway. If the gateway
allows re-invite than the rtp would go directly from phone A to the
gateway, while the sip messages are still going through Asterisk.
I have problems with pickupgroup. While 621 can pickup a call to 601
with *8, no phone can pickup a call to 621. Below are the settings for
two phones. 601 is static in the sip.conf, while 621 is in the Real-time
database. What could be the problem?
I have an extension 601:
[601]
type=friend
I want to change that ASTCC will ask for pin.
1. Where to set it? Pin length and number?
2. Can I set the pin only for a few people? E.g. Would deleting the
pin number not ask for the pin or needs than still the #
3. How to change the pin? Can the user change the pin?
bye
I need h.264 and tried therefore svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk
(currently I have branches 1.2 installed)
make clean; make update; make install
.
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk'
rm -f .depend
rm -f .depend
rm -f .depend
I see in CLI:
ast_parse_allow_disallow: Cannot allow unknown format 'h264'
What can I do ?
I see on Asterisk home page, that h264 is not listed.
When does Asterisk need h264 at all? If one phone calls another phone,
than it is only passed through and does not need it, or am I wrong here?
My last update was a while back and as I remember svn trunk did not
compile and I was advised to use branches 1.2 till further notice.
Have I missed the further notice and can we use now svn trunk or is the
advice still to use branches 1.2 ???
bye
Ronald
I thought with iptable -L |grep udp I will find out which ports are
open for the rtp stream, but I cannot get this info from here, or
at least I cannot interpret it:
# iptables -L |grep udp
ACCEPT udp -- anywhere anywherestate
RELATED,ESTABLISHED
LOG
I need somebody who can test with me video phone settings.
I use Eyebeam!
Please contact me via MSN first: [EMAIL PROTECTED]
bye
Ronald
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Koopmann, Jan-Peter wrote:
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?
Because the transfer button
Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits
Koopmann, Jan-Peter wrote:
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
try that way. However, I have
,
bye
Ronald
HTH
routerguy
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits
have different length and
overlapping,
bye
Ronald
CP
On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:
I found a problem in blind transfer:
I have an extension number 601 and I have an extension 6014
If I get a call on 615 (snom) and transfer to 6014 it works, since snom
requires
.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits
Anthony Rodgers wrote:
With respect, the problem
/asterisk-users
---
avast! Antivirus: Inbound message clean.
Virus Database (VPS): 0635-4, 2006/09/01
Tested on: 2006/9/2 ¤U¤È 04:42:02
avast! - copyright (c) 1988-2006 ALWIL Software.
http://www.avast.com
--
Ronald
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Saturday, September 02, 2006 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Keys pressed not registering ...
Lenny wrote:
Hello all,
For some reason when
Ronald Wiplinger wrote:
David Gagnon wrote:
Ronald,
You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?
David
David,
I am not angry about VoIP, but please send my your old Nortel system
!
I just do
count?
bye
Ronald
-Tim
On September 2, 2006 20:12, Ronald Wiplinger wrote:
Kevin Smith wrote:
Dialing a number and transferring a number are two different things.
And no offense, you are not really providing a lot of details along
with your problem. So you can dial the numbers
I found a problem in blind transfer:
I have an extension number 601 and I have an extension 6014
If I get a call on 615 (snom) and transfer to 6014 it works, since snom
requires me to hit ok
If I get a call on 601 and transfer to 6014, than 601 will get the busy
signal and I hang up as
[Aug 31 04:32:22] NOTICE[20241]: chan_sip.c:5291 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying
again (Attempt #984)
[Aug 31 04:32:23] NOTICE[20241]: chan_sip.c:9600
handle_response_register: Failed to authenticate on REGISTER to
'[EMAIL PROTECTED]' (Tries 3)
I cannot explain why I get all the time:
Got SIP response 486 Busy Here back from 192.168.250.244
I have a Wellgate 3804a there.
How can I solve it?
bye
Ronald
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I want to record a call, either it is an incoming call or an outgoing call.
I have in features.conf:
automon = *1
However, I am not sure if that is what I need, and how to use it.
bye
Ronald
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Michael Welter wrote:
My Asterisk colo server has been up for almost two years. Today it
crashed. When I gave the reboot command, it crashed so hard that it
had to be power cycled. I wasn't in attendance, but I can speculate
that it had a kernel panic during the shutdown.
Yesterday I
Tim St. Pierre wrote:
Try creating an extension with a lower priority that answers the channel
first. If you don't, the application will run, but the call will timeout as
no answer, since it was never actually answered. It sounds weird, but this
is how you get messages like please check the
I want that each call from PSTN goes to Asterisk to the context for this
line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
My sip.conf:
[WG88621001]
type=friend
defaultip=192.168.250.244
insecure=very
I am looking for a way to make a decission in the dialplan if I have a
caller id or not.
What I want to do with it:
Call on the PSTN line should either use astcc.agi with the caller-id in
place as card number, or asking for the calling card number.
How can I make this gotoif ???
bye
My dialplan grew over the last months and I want to restructure it. What
hints do you have for me?
There are some points I want to do, but none of my tests worked.
I use realtime, and have there a field called key, which can have
several flags.
E.g. a flag if the user is allowed to use a
sox needs for gsm an optional library.
I was not able to locate this one. Can anybody point me to this place?
bye
Ronald
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Is there a table available, which tells me if a zip code, city and area
code matches?
For now I did it with google, type each info in and found out if it
matches, but it would be easier if there is a table available.
bye
Ronald
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Kevin P. Fleming wrote:
Can we please keep the discussions about carriers, money, jobs, work, etc. off
of this list? This is not the place to discuss your experiences with _any_
company, it's a place to talk about Asterisk and using Asterisk.
Please move flamewars and similar discussions to
/12 ¤W¤È 09:27:28
avast! - copyright (c) 1988-2006 ALWIL Software.
http://www.avast.com
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com
Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208
- I'm a SpamCon
To keep the Asterisk mailing list free of Voip provider complaints:
VoIP is a growing business area. We all find days of problems. Some
companies can handle problems. Some VoIP providers create problems. In
this group we can discuss and learn how to handle conflicts.
What to do and what not
Dear NuFone,
Without misunderstanding I ask you again, please send the log file and
pay back my money!
Not following this request results in the assumption that NuFone is
cheating and I will post this info every hour on more Internet places.
This should help that other people will not trap
Andrew D Kirch wrote:
Ronald Wiplinger wrote:
Dear NuFone,
Without misunderstanding I ask you again, please send the log file
and pay back my money!
Not following this request results in the assumption that NuFone is
cheating and I will post this info every hour on more Internet places
Andrew D Kirch wrote:
Ronald Wiplinger wrote:
Dear NuFone,
Without misunderstanding I ask you again, please send the log file
and pay back my money!
Not following this request results in the assumption that NuFone is
cheating and I will post this info every hour on more Internet places
Andrew D Kirch wrote:
Michael Workman wrote:
So Nufone Screwed ya
I feel Sorry... W Take your Lumps... Cut Your Losses and Get
on with
Life
Your not the only one Nufone Screwed They Screwed me Out of
$3,000.00
How do you figure this at 2.9c/min?
Andrew
That is
Michael Workman wrote:
I am not talk about Call Time.. They Screwed me by Me Hiring them to consult
on setting up server and they took the money and never did the work
This they tried also with me, but I only answered, that I would like to
learn it by myself,
bye
Ronald
Part of a conversation with NuFone.
It is untrue, that they do not answer, but if than:
Quote:
3. change your attitude towards customers!!
No, if you don't like it, go use Vonage.
End of quote!
I had always problems with these people.
bye
Ronald
The big player show us, to limit the free phone calls per week to a
certain amount.
How can we do that with ASTCC?
bye
Ronald
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I have patched astcc.agi with the HUP patch, but it still hangs from
time to time.
Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running
Linux on 2006-05-07 00:31:09 UTC
bye
Ronald
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I want to get a variable, depending on the time.
I tried this one, but it does not work:
exten = 75,1,Set(guess=SYSTEM(echo $((1 + $(date +%S)*100 % 23)))
The idea is that the variable guess will change every 23 times per minute.
How would be the right syntax?
bye
Ronald Wiplinger
long phone calls are, separated to different criteria, like
prefix number, duration.
most of these is in the program from areski, with the exeption that the
numbers are wrong, like graphic shows 5 phone call and load shows 4
calls, .
What are you using?
bye
Ronald Wiplinger
done.
After that, I want to go to the next include, which has a match for
_91NNN.
However, since the first match was already successful, the next includes
are not visited anymore.
How can I overcome this problem?
bye
Ronald Wiplinger
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when I try
asterisk -rx show channels concise
I get an output of:
SIP/tf.voipmich.com-8671 ...
SIP/1110-78ac
The phone 1110 is not anymore on a phone call. How can I remove this
zombie channel?
bye
Ronald Wiplinger
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?
bye
Ronald Wiplinger
[mycontext]
include = var-key
include =
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I cannot explain that:
One of my users shows up in sip show peers as 654200/Elmit_Unl
I can set it back to 654200/654200 but it will change back to
654200/Elmit_Unl
Why?
bye
Ronald Wiplinger
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How can I use the column setvar in my dialplan?
I am not sure if it is for that what I need:
Many phones have the same jump in place, but need a few variables
different, like tariff, silent, need_password,
I have for tariff = 4 variations, for silent=2, for need_password=2
... If I solve it
JP Carballo wrote:
Ronald Wiplinger wrote:
I got a request for one customers to set-up 100 accounts.
I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?
To generate 100 cards is not a problem, but if it would work with one
account number would
context for this customer and use only his
account code as card number.
Any advice would be appreciated.
bye
Ronald Wiplinger
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problem will be fixed then.
Regards,
Can you please give us more info about that?
What is php-pcntl? What should it do? How can it be used to be a solution?
bye
Ronald Wiplinger
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JP Carballo wrote:
Ronald Wiplinger wrote:
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are
many other cases.
I have created a number the user can dial to reset this flag.
However, that is written
want to make to reset all in use flag with a program. Has anybody done
it, or has a better idea?
My idea is to check every 5 minutes, the database, which cards are set
in use and check if this is true, if not reset it.
Q: How do I know if a card is in use?
bye
Ronald Wiplinger
Is anybody using XEN in conjunction with Asterisk and/or Openser?
I would like to get some info about such an environment and experience
reports.
bye
Ronald Wiplinger
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Betamax makes our life more and more difficulty, hehehehehehe.
I found (today) that the free calls are limited to 300 minutes per week.
It is good to know what excess use means!
That gives now also a challenge in the dialplan
Let's assume we have 5 accounts, each one has 300 minutes.
We
I failed to transmit dtmf via voipbuster to the destination.
Does anybody have success, if how to set it up?
bye
Ronald Wiplinger
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* *0
Dial option is tTwWr
I tried to call from 601 to 615
601 keys in *0nothing happens
603 keys *8 I get the phone call
# 621 nothing happens
What do I miss ???
bye
Ronald Wiplinger
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I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?
exten = 8863959,1,Dial(SIP/8863959,60,r)
exten = 8863959,2,NoOp(${DIALSTATUS})
exten = 8863959,3,Voicemail,[EMAIL PROTECTED]
exten = 8863959,104,Voicemail,[EMAIL PROTECTED]
exten
Can anybody tell me how I can key in # codes after the call is established?
All what happens now is that the call will be placed on hold and a new
call will initiate!!!
bye
Ronald Wiplinger
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Ronald Wiplinger wrote:
Can anybody tell me how I can key in # codes after the call is
established?
All what happens now is that the call will be placed on hold and a new
call will initiate!!!
Just enter the required digits, just as if you are accessing voicemail.
Don't press the send
Discussion
than you may specify your problem so that we can try to help you.
Just wanted to warn you guys
wow!
bye
Ronald Wiplinger
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I would like to test my extensions.conf before I give it to my users.
Is there a dialplan emulator available?
bye
Ronald Wiplinger
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I upgraded recently to Asterisk SVN-branch-1.2-r25165M
the commandline
asterisk -rx 'sip show peers'
returns with the first line:
on
Is that a bug, or how can I omit it?
I used:
asterisk -rx 'sip show peers'|grep OK|sort | tee /dev/tty |wc -l; echo
registered at ELMIT
which results
the same! Could anybody solve it?
bye
Ronald Wiplinger
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I upgraded * via svn and it did not work !!!
1. asterisk-addon did not compile!
pbx:/usr/local/src/svn-versions/asterisk-addons # make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
make -C format_mp3 all
make[1]: Entering directory
If I call PSTN number a, than I can call the extension number, while
when I call PSTN phone number b the tones are ignored.
If I call PSTN PSTN directly the extension number can be dialed.
How can I improve that?
bye
Ronald Wiplinger
How can I install a softphone on my USB flash drive like Xlite and have
it ready to go when I plug it in at any Windows XP computer?
(Same for a Linux softphone, both on one USB flash drive).
bye
Ronald Wiplinger
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should take care of the rest.
It is a nice phone, but it is IAX. I would like to use a SIP phone.
The reason for that is that there is no IAX server for the mass, but
openSER
bye
Ronald Wiplinger
On 5/1/06, *Time Bandit* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
How
- the newly
parent company of skype, for a not received parcel, but the rules says,
below 25 US$ there is no guarantee that you get anything
bye
Ronald Wiplinger
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Requirements
* Windows 98/2000/Me/XP/2003
sigh
bye
Ronald Wiplinger
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