Hello Matt,
very interesting setup! are you using asteriak queues for inbound or not
at all?
Bye
l.
In data Thu, 22 Sep 2005 06:25:44 +0200, Matt Florell [EMAIL PROTECTED]
ha scritto:
We wrote VICIDIAL(part of the GPL astGUIclient suite
http://astguiclient.sf.net) for our call center
Oh yeah,
And:
Turn OFF Filter Packets from Registrar.
Turn ON Support Broken Registrar.
This may or may not be a security risk, but for testing, it will help to see if
toggling these make a difference.
Shanon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Michiel van Baak wrote:
Set(CALLERID(number)=0123456789)
SetCIDNum(0123456789)
And
Set(CALLERID(name)=My name)
SetCIDName(My Name)
I prefer the old one, was much more simple. It seems with the new formats in
Asterisk I always end up with extension lines that are wider than my screen
width
Hello,
i wonder why i didn't find a solution for this problem yet, because it
seems very common:
I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some
SIP-Softphones which i can call from outside by calling the phonenumber of the
Asterisk-Server and then dialing the number of
Okay here is a quick breakdown of my situation; I have an
asterisk server with AMP installed. Amp stores all of the CDRS in a mysql
database and comes with a nifty web based reporting tool that has worked well
for me. The problem I am running into is that my mysql database is
nearing 2
Greetings to all,
Im try to compile * (1.2) on a Sunfire 210 with Solaris 10, but
do not get past line 29 in the Makefile. Some innoccuous line with uname
s as a variable.
Would love to hear from anyone who has gotten Asterisk to compile on Solaris
10 specifically.
Thanskt to
Chris Modesitt wrote:
I am currently reprogramming the msyql database to only store 30 days
worth of information and archive the rest. This will get rid of the
majority of our symptoms however the underlying problem is that the
mysql/php/httpd daemons can saturate the CPU and starve
On Wed, Sep 21, 2005 at 09:44:01PM -0700, Pradeepa Ramamurthy wrote:
I have installed Asterisk and i have configured with two
SJPhones; i am able to make calls between these two phones.I am planning to
develop a application basically web based
application
from which the administrator able
On Thu, Sep 22, 2005 at 09:29:20AM +0200, Joseph Rothstein wrote:
Greetings to all,
I'm try to compile * (1.2) on a Sunfire 210 with Solaris 10, but do not get
past line 29 in the Makefile. Some innoccuous line with 'uname -s' as a
variable.
Anybody tried using INSTALL_PREFIX with a
On Thu, 22 Sep 2005 [EMAIL PROTECTED] wrote:
Hello,
i wonder why i didn't find a solution for this problem yet, because it
seems very common:
I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some
SIP-Softphones which i can call from outside by calling the phonenumber of
Use
${CHANNEL} to get the number
then
SetCIDNum() to set the new number w/ zero.
Tomasz Chmielewski wrote:
I have an
asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that
Which version of the driver do you use?
Fernando Herrera wrote:
Hello,
I have installed oh323 channel driver. Outgoing calls to H.323 world do
not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that
userInputMode=RFC2833 has already been set.
Does anyone know how to make
Alchaemist Wrote:
Now... if you have dynamic IP in the asterisk... things change
because
Asterisk must know in sip.conf the external IP.
I think I read in this list, that the best (only?) way to get arround,
is to
place a script that detects the external IP when it changes,updates
sip.conf
I'm having
problems updating too.
The beta 1.0.7.11 release has fewer files than the 1.0.6., so tried the
following:
copied the 1.0.6.x files the the new
directory, then uncompressed the 1.0.7.11
bundle in the new directory. this effectively overwrites the files with
the same name.
Hello,
Try insecure=very in [sip.philonline.com]
Harry
--- Ryan Pagquil [EMAIL PROTECTED] a écrit :
Hi,
I'm setting up Asterisk as a voicemail with
SER. My problem is,
when a caller that is not registered with asterisk
(no username and
password in sip.conf) it prompts 403,
Correct: Use ${DNID} to get the number. I'm sorry.
Bruno.
Bruno De Luca wrote:
Use
${CHANNEL} to get the number
then
SetCIDNum() to set the new number w/ zero.
Tomasz Chmielewski wrote:
I have an
asterisk box and SIP / IAX2 phones.
To call out, users have
Hi :)
I hope someone has a hint concerning Early Media.
The situation:
My Asterisk is connected to small local carrier who works with several SIP
servers.
I traced some SIP headers and find something like this:
Via: SIP/2.0 UDP sip1.provider1.de
In the SDP part I found something like this:
o=-
Bruno De Luca wrote:
Correct: Use ${DNID} to get the number. I'm sorry.
Bruno.
${CALLERIDNUM} is the most commonly used.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php
I'm hosting soft-switch.org now - Steve has said he doesn't want FTP, so it's
all http now. Feel free to update the wiki.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, 22 September 2005 11:50 AM
To: 'Asterisk Users
[EMAIL PROTECTED] wrote:
Now, I traced RTP packets and see how sip2.provider1.de sends
packets to my Asterisk but the port seems closed on my server so the
inquiring server of
provider1 will never get an answer and sends a port unreachable.
Did provider1 send the exact same SIP message types
Hi :)
Am Donnerstag, 22. September 2005 12:48 schrieb Andreas Sikkema:
[EMAIL PROTECTED] wrote:
Now, I traced RTP packets and see how sip2.provider1.de sends
packets to my Asterisk but the port seems closed on my server so the
inquiring server of
provider1 will never get an answer and
This is a limitation of your PSTN provider.
The telco's don't allow you to set your callerid number when dialing out.
They always change it to the one that is allocated to you.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL
Hello,
I would like use some IAX client for Linux text console. I am blind, so
I can not use some X client in this moment. Knows someone about some
possibility?
Thanks.
--
Jan Buchal
Tel: (00420) 224921679
Mob: (00420) 608023021
___
--Bandwidth
Slight correction... _some_ pstn providers do allow one to set the
callerid number (but not the callerid name). Its an option they can
turn on/off, but depends on the provider's policy.
This is a limitation of your PSTN provider.
The telco's don't allow you to set your
I hope the subject isn't too buzzword compliant :)
I'm just curious: What have people done with Asterisk? I'm particularly
interested in DIY projects and things that can be done on a small/home
office (or even hobbiest's) budget. If you have clever hacks or
creative functionality you've
2005/9/22, Jan Buchal [EMAIL PROTECTED]:
Hello,
I would like use some IAX client for Linux text console. I am blind, so
I can not use some X client in this moment. Knows someone about some
possibility?
Asterisk itself, if properly configured, can work pretty well as an
IAX client. Using the
How do I combined these in and out wav files on the
fly through asterisk to where I hear the whole
conversation and only have one wav-file
(i.e. :
agent-1001-asterisk-478-1127389080-17-in_out.wav)
agent-1001-asterisk-478-1127389080-17-in.wav
agent-1001-asterisk-478-1127389080-17-out.wav
Hi all,
I'm trying to connect my [EMAIL PROTECTED] to iptel.org, but the only I
get is Allison telling me circuit busy now, please call again later
or some thing similar.
I'm trying make it by AMP and editing sip.conf and extension.conf, and
I read all about it in voip-info.org.
I will
I know this has been discussed heavily but i have a
bizzare issue with call pickup.
I have 3 asterisk servers all built the same on centos
4.1 and call pickup works on two of them but not on
the third. They have identical configurations.
I'm using asterisk 1-0-9 and zaptel with ztdummy. All
Hello Asteriskers,
We are proud to announce, our initial support for AMPortal in Debian
Sarge: we are releasing Debian packages containing AMPortal 1.10.008,
partially working under our Rapid distribution, the packages should also
run on normal Debian Sarge and hopefully also under Sid, Etch,
sip debug?
- Original Message -
From: Sebastian Milioto [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 22, 2005 6:18 AM
Subject: [Asterisk-Users] Asterisk with iptel.org
Hi all,
I'm trying to
Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where to configure this.
thx.
--- Jacky [EMAIL PROTECTED] wrote:
LCS 2005 just support SIP TCP or
Hi list. I´m using ASTCC with callerid authentication and got things
working fine, except for one single issue:
Using this command -- DeadAGI(astcc.agi|${CALLERIDNUM}|${EXTEN:1}|5) --
and passing the 5 parameter for silent, it exits unexpectedly. I tested
the 4, 3, 2 and 1 and they are
I have another freaky situation that is occuring. I am getting
overlapping phone calls on the same phone.
Basically when a user hits the hook fast enough, the current SIP phone
call connection does not get killed by the Asterisk server while a
second SIP connection gets created giving the user
Ill update the wiki whenever I stumble into ftp mentioned again.
Also the links from Steves page will need to be updated.
Thx for the info Rob.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rob Thomas
|Sent: Jueves, 22 de Septiembre de 2005
On Sep 21, 2005, at 1:46 AM, Zoa wrote:
True... But i tried several brands of cards, and several drivers, the
dual nic gigabit intel card was a lot better than all the other
combinations i tried.
Fun fact: short of buying a 10gigE card, your best performance will
probably be from a cheap
Have you discovered www.voip-info.org yet?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian McEntire
Sent: Thursday, September 22, 2005
6:18 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SOHO
Survey / Creative Asterisk Solutions
I
Good afternoon,
I have to develop an Asterisk system in Spain that
can support up to 8 simultaneos ISDN calls (4 BRI), initially only 6 simultaneos
calls.
I have been looking into VOIP-INFO and Ifound
that the Junghanns card QuadBRI is a very good option. But when I look for a
server to
Well guys here comes the fun part. I have a Microsoft access
(VBA) application that interfaces with my SQL database. This app pulls of info from
the SQL record and than picks up the phone and dials that locations number. I
have purchased a few hundred NpaNxxs for my own use. I want get
Hi,
I plan, if I have time, to work on it this weekend, so in that case
I will post the code, no problem at all.
I want to add reply with record functionality first.
Regards!
Alchaemist
Michiel van Baak [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On
lol
just posted this yesterday, it's for any ODBC DSN so Access or SQL or an Excel
spreadsheet, as long as it's set as a DSN. This will work with outgoing Caller
ID as well, it's just how you set it up in your dialplan. If you want I can
email you the .agi since email will undoubtedly
Tim King wrote:
Well guys here comes the fun part. I have a Microsoft access (VBA)
application that interfaces with my SQL database. This app pulls of
info from the SQL record and than picks up the phone and dials that
locations number. I have purchased a few hundred NpaNxx’s for my own
use.
Few weeks back local telco introduced option of custom ring tones. I am
not talking about your phone ring tones but about ring tones you hear in
your headset while phone is ringing on the other end.
If I understand correctly, ringing tone is generated localy on asterisk
if you are connected to
ACCESS supports ODBC driven connections..
Well guys here comes the fun part. I have a Microsoft access (VBA)
application that interfaces with my SQL database. This app pulls of info
from the SQL record and than picks up the phone and dials that locations
number. I have purchased a few
If you used a perl or PHP agi script, you could probably use some kind
of ODBC drivers to communicate between the two.
--
Tom Hayden
On 9/22/05, Tim King [EMAIL PROTECTED] wrote:
Well guys here comes the fun part. I have a Microsoft access (VBA)
application that interfaces with my SQL
I have a need to use
cdr_custom and would like to know if anyone has gotten it to work with a mysql
cdr backend, and any examples if possible
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Hello,
If I change the password in the Voicemail-Menu without externpass, the
password gets changed correctly. But if I use a extern script, the password
is still the same as before.
What do I have to do in my external script to change the password in the
asterisk's memory, too?
Thanks for your
Hi,
Lets see... dynamic IP, means mainly two options:
1- PPPoE in the same machine as asterisk
In that case, you can get the IP locallyin the shell
2- Whatever protocol, in a router
In that case you must rely in querying your router or an external system
(like www.myipaddress.com)
Yes, and it is a fantastic resource! I don't think I could have gotten up and running without it.
I assume you pointed me there because there are some articles about
what people are doing with *. I'll dig for them. I previously read in
an issue of ;Login: several columns discussing things you
With sip show registry I can see I my asterisk is registered:
Host: iptel.org:5060
Username:84565616
Refresh: 145
State: Registered
The following is part of sip.conf:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on
I seem to recall this problem on the mailing list a couple of months ago,
I'd point you towards, but can't seem to find it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Braz
Sent: September 21, 2005 9:29 PM
To: asterisk-users@lists.digium.com
Please forgive the intrusion ...
We have a Sangoma A102u dual-span T1/E1/J1 card coming available as we are
upgrading our equipment. We have the original packaging, cables, manuals and
software that came with this card. It has only been used for 3 months
in a data
center environment and is in
Yeah. It's a brilliant idea because I believe they would probably return
answer supervision to play these custom ring tones therefore creating
more revenue from the incoming calls.
Marko Rakar wrote:
Few weeks back local telco introduced option of custom ring tones. I am
not talking about
Hi,
Dial application with m option, if the telco accept from you.
On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote:
Few weeks back local telco introduced option of custom ring tones. I am
not talking about your phone ring tones but about ring tones you hear in
your headset while phone is
Hi,
In my dialplan I'm using a WaitExten() command. It works only with Zap
phones. When I dial this command with Sip phone asterisk do nothing.
Should I put extra definition in sip.conf to make this work with Sip
phones?
Thanks in advance
Cheers
___
try [EMAIL PROTECTED] it has some builtin features like the weather script
and stuff and if you are just playing around and have fun then this
would be the way to get some nice features and such.On 9/22/05, Brian McEntire [EMAIL PROTECTED]
wrote:Yes, and it is a fantastic resource! I don't think
Hi,
I just got my ip500 back after months of waiting. Is there an easy way
to get it hooked up to asterisk without [t]ftp servers and all that or
is there a quickstart page somewhere?
tia
r
___
--Bandwidth and Colocation sponsored by Easynews.com --
yes, but I want this feature to be turned on for people who are calling
my asterisk from PSTN
Two atoms bump into each other. One says
I think I lost an electron! The other
asks, Are you sure?, to which the
first replies, I'm positive.
mailto:[EMAIL PROTECTED]
http://printel.hr
Try forcing the dtmf mode, such as
Exten=EXTEN,1,SipDTMFMode(inband)
That worked for me, but you'll need to only do it on the SIP calls, so route
accordingly
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Andrew Nowrot
-Sent: Thursday, September
similar service is when you call our local telco customer service, when
you dial it then you first hear you have reached customer service,
plase wait and while you hear that your call is still not connected and
therefore it is free
in addition you might have similar message to the caller that
Dear All,
I have tested more than 4 windows clients to asterisk
I am testing them remotely through adsl at the client and adsl at the
sever.
there is a big problem in the QOS from * to the client is OK while from
the client to * is very . I have investigated the problem using
ethereal . I
On Thu, Sep 22, 2005 at 02:25:02PM -0300, Sebastian Kühner wrote:
Hello,
If I change the password in the Voicemail-Menu without externpass, the
password gets changed correctly. But if I use a extern script, the password
is still the same as before.
What do I have to do in my external
It's best for you to set up an ftp server instead of a tftp server, but
I don't think you'll enjoy setting up a soundpoint phone without either
of them. The Polycom Phones page in the wiki was pretty much all I
needed to set mine up:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
make sure the program 'sox' is installed, the in and out streams get
muxed automatically (not quite on the fly but after recording has stopped)
Crystal Stream, Incorporated wrote:
How do I combined these in and out wav files on the
fly through asterisk to where I hear the whole
conversation
Joerg Lauer schrieb:
Hi,
I think I had the same problem and I think the error was that the dial
statement had to be:
exten = s,1,Dial(Zap/1/,60,tT)
I may remeber wrong, though.
Btw: It may be a better idea to use Zap/g2/,60,tT), this way both
B-Channels of the HFC card may be used.
Huh? You can easily configure an IP500 via a web browser. Just point
the URL to the IP addr of the telephone.
--
Tom Hayden
On 9/22/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Hi,
I just got my ip500 back after months of waiting. Is there an easy way
to get it hooked up to asterisk without
You need to run a late 12.3T to get name passed via SIP.
Max Braz wrote:
Hi guys.
We have currently Asterisk CVS-v1-0-08/15/05-15:53:48
connected in SIP with a Cisco AS5300 (IOS 12.3). One
PRI is connected to the Cisco gateway.
The problem we have is that on incoming PSTN calls to
the
In short: use the right tool for the joob. See below.
On Wed, Sep 21, 2005 at 05:34:23PM -0300, Alchaemist wrote:
Hi,
Has anybody seen a non commercial, or freeware, or GPL, or even
CHEAP... POP/IMAP to Text-to-speech?
Pull pop3/imap/imaps/whatever with fetchmail (or getmail,
but you do not get all of the features via a web browser to customize.On 9/22/05, Tom Hayden [EMAIL PROTECTED]
wrote:Huh? You can easily configure an IP500 via a web browser. Just pointthe URL to the IP addr of the telephone.
--Tom HaydenOn 9/22/05, Wilson Pickett [EMAIL PROTECTED] wrote: Hi, I
i have registered on teliax service and i m using a hathway internet connection.with X-lite phone it is not logging in .it says login timed out whereas the phone with same X-lite and service settings i m getting logged in all other internet connection and the phone also works perfectly. the
Hi,
Does anyone know if the Digium Wildcard will work on a PCI Express or
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack
server for use with Asterisk.
Thanks
___
--Bandwidth and Colocation sponsored by Easynews.com --
Dial(SIP/1234|90|m)
the caller will hear music on hold while SIP/1234 is ringing
On Thu, 2005-09-22 at 20:18 +0200, Marko Rakar wrote:
yes, but I want this feature to be turned on for people who are calling
my asterisk from PSTN
Two atoms bump into each other. One says
I think I
Chuck Bunn wrote:
Does anyone know if the Digium Wildcard will work on a PCI Express or
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack
server for use with Asterisk.
They will work in PCI-X of course but not PCI Express. They are
totally different.
You will need the
Look at the dial app. I think it has several options.
Most custom 'TONES' are wav, acc, mp3 etc. files.
If you can set a different MOH class or perhaps a playback file in the dial
app that plays a file that is a 'RING TONE' that may work.
-John
-Original Message-
From: [EMAIL
Try putting the command timers buffer-invite 5000 in the sip-ua config.
This works on both our 3640 and 7206. I'm not sure if this command is
available in the 12.3 series as I have 12.3T on my equipment.
B. J.
-Original Message-
From: Max Braz [mailto:[EMAIL PROTECTED]
Sent:
I am not interested in Dial app, I want the callers who are calling FROM
pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or
whatever)
For users within asterisk domain who actually use Dial command it does
not matter and I know that I can have full control over them
Two
Bump!
raj
Rajkumar S wrote:
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the
Hi Matt,Is your solution 100% Asterisk or are you using other "helpers" such as SER or XXXproxy or whatever?Thanks,WaldoOn Sep 21, 2005, at 12:45 PM, Matt Roth wrote:All, This message has generated a lot of responses, so I'm going to address each of them here in an attempt to consolidate the
Marko Rakar wrote:
I am not interested in Dial app, I want the callers who are calling FROM pstn
TO asterisk to hear different kind of ring tone (wav, mp3, gsm or whatever)
??
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Then
I think Dial will work for you too, though: In your incoming from pstn
context, Answer, then Dial providing music on hold or Backgrounding an
audio file.
I assume, however, you don't want to answer the line at all. You don't
want the remote caller to be billed for this call, and you just
After dealing with a Poly 301 I rather use the FTP server and config
files, even for a single phone,
download the manual and stuff from freedomphones.net/polycom
Tom Vile wrote:
but you do not get all of the features via a web browser to customize.
On 9/22/05, *Tom Hayden* [EMAIL PROTECTED]
Excellent!!!
Thanks fellas.
Mark
David Mallwitz wrote:
Mark Phillips wrote:
I was at VON in Boston today and saw on the Digium stand a Cisco 7960
with a picture of Tux and the Asterisk log on its display. I WANT IT!
Anyone know where I can download this file please?
yes, yes
the thing is that local telco uses this feature for their customer
support line and also one of wireless providers now also offers ability
to customize your ring tone
I was told that if you have analog or even ISDN BRI line that ring tone
is generated in your local teclo exchange, but
??
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Then you go to IVR, VM or ??
OP said T1/E1, so (usually) there's no ringing delay until Asterisk picks up
like in a POTS line. Answering a T1/PRI line is transparent to the caller
and then
We are 100% Asterisk on the VOIP side. We use SIP, IAX and Zap(channelbanks) for phones and Zap T1s for telco termination.
MATT---On 9/22/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hi Matt,Is your solution 100% Asterisk or are you using other helpers such as SER or XXXproxy or
John,
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Generally the ringback tone is sent by the last ClassV/Class IV switch in
the telephony path. This is for Telco's to send inband
error/progress/information announcements. However,
I have a [pstn-inbound] that calls a dial app that plays music to the pstn
caller.
--john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Marko Rakar
Sent: Thursday, September 22, 2005 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial
Forgot. You must answer the line first.
Other than that Asterisk is not involved with the external pstn until it is
answered.
--john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Marko Rakar
Sent: Thursday, September 22, 2005 2:31 PM
To:
Don't bank on it. We were going to use a Wildcard as a timing source on
our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the
PCI-X slot I installed it in, sometimes the box wouldn't even boot. For
perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit
133 MHz,
On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote:
yes, yes
the thing is that local telco uses this feature for their customer
support line and also one of wireless providers now also offers ability
to customize your ring tone
I was told that if you have analog or even ISDN BRI line
Just correcting myself. The 3 PCI-X slots are one 64-bit 133 MHz and
two 64-bit 100 MHz.
Matt
Matt Roth wrote:
Don't bank on it. We were going to use a Wildcard as a timing source
on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on
the PCI-X slot I installed it in,
I was thinking of PSTN over FXO cards. When I see PSTN I think pots.
You mentioned BRI whould PRI do as well?
--john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Fernando Herrera
Sent: Thursday, September 22, 2005 3:07 PM
To: [EMAIL
On Thu, 2005-09-22 at 14:02 -0600, Colin Anderson wrote:
??
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Then you go to IVR, VM or ??
OP said T1/E1, so (usually) there's no ringing delay until Asterisk picks up
like in a POTS
Hello,
I found another solution - a script that does the following steps:
1.
Get the old password from
the db
2.
Update database
(subscriberàpassword) with new password (that's for my web
interface and ser)
3.
Replace old with new
password in the file
Hi,
is there a way to set the log level to the equivalent of:
asterisk -vdc
Or anything similar? I have I problem I cannot really trace with the
standard log level.
Thanks,
Arik
___
--Bandwidth and Colocation sponsored by Easynews.com --
If you use FXO you must answer before sending media. If you use BRI, I think
you should be able to send media after sending Progress/Alerting.
But, watch out!! As trixter said in a previous mail
Check your local laws on that
Local regulations may limit the amount of information you can send
On Thu, 2005-09-22 at 22:33 +0200, Arik Funke wrote:
is there a way to set the log level to the equivalent of:
asterisk -vdc
Or anything similar? I have I problem I cannot really trace with the
standard log level.
Have you looked at logger.conf and the full variable?
--
Jesse
Guys.
Im having some audio problems.
My asterisk box is the router for my network but remote softphone cant
connect to it. I tried running 1.2beta1 and cvs-head with the same luck and
etheral throws this:
577.358145 201.138.93.80 - 201.129.249.28 SIP Request: REGISTER
Hello,
I have not been following the asterisk list the last few weeks but from
the number of requests I have received asking this question I guess it
is best to send my answer to the list for reference.
So here is what I did to compile bristuff on Fedora Core 4.
Make the following changes to
- The reason i recommended you to use a ramdisk is because i think the
- problem with recording to disk is saving 20ms of stream 1, then 20 ms of
- stream 2, then 20ms of stream 3 etc etc meaning you write everytime
- very small things. (with a lot of seeking).
I was thinking about
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