Hello there
I'm quite a newbie in the IP Telephony area. I'm playing a little around
with a setup with one linux box with a e100 p card installed, which
functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper).
I have two h323 phones, Welltech WellGate 1501 and 3502.
So far I've managed
I am still poking at app_festival a little bit and have found a problem
I don't really understand, and therefore, I don't really know what to
try to do about fixing it. I have done a lot of things in the code, but
the best I can do is rewrite a bunch of stuff and still have the
problem. Perhaps
Hello guys,
Have you guys experienced problem with digit 0 from x-lite.
I cannot access asterisk services with number containing
digit 0. For example i have voicemail with mailbox number
6100, if i login from x-lite login always fail because
asterisk only read number 61.
Hi,
I am a new to Asterisk. I am looking for a cheap solution for PC-to-Phone
call serving one or two cities. Can anyone provide me with pointers to
architecture documents/other documents from where I can start. I am not
new to VoIP.
Regards,
Deepak Mittal
it works fine with me, i tried 6100 and the phone call starts without
any problem...
i don't know what is happening in your configuration...
could be the problem not related to your x-lite?
try to see the trace of asterisk and look at the number that asterisk
process when you make a call
Angelo
Hi!
I've installed Asterisk and connected ATA-186. When I press 8500, I
listen voice main menu and prompt for enter mailbox number. I press 1234,
but asterisk not accept number and switch to demo-instruct.
Also Asterisk write warning:
NOTICE[77839]: File rtp.c, Line 221 (process_rfc3389):
hello,
I have a test setup with 2 asterisk servers, each having a one snom 100
via sip using it. I`m experimenting on how trunking between them would
work. I have them setup for RSA authentication which I plan to use in
the future.
So I`ve setup the keys and servers seem authenticate to each
Hi,
I have the same error only whan I start Asterisk with asterisk -vc
If I start it with safe_asterisk and then enter to the console with
asterisk -vvr this message does not appear.
Any type of calls I made from the SIP phone I get the same NOTICE type
message.
Anyway, it seems to work
as a follow up: when I make a call to the extension on the other box
designated as switch, I see packets going to to iax port on the switch
box but I dont see any relies from it.
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
___
RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389
packets. You can turn this off through the ATA 186 web interface.
It looks as though you need to configure that ATA186 properly - several
people have posted guides on this.
Iain
--On Thursday, July 3, 2003 9:29 am
Hi,
I have installed an USB modem and test it using the strings from the
included Aopen modem (Rockwell based), modified the modem.conf file and then
Asterisk load them without errors.
What is the syntax to be used with a modem as a FXO device when dialing?
I have tried:
exten =
Hi,
Just got a couple of Bugetone's yesterday..
I updated the firmware to 1.0.3.72 but now it seems to have problems getting the time
from the default NTP server.. I also tried time-b.nist.gov but still the time does not
seem to work..
I am able to contact the time server from my PC..
Anyone
Hi Iain,
Do you know how can I disable this on Cisco 7960?
There is no such an option in the menu, nor in the configuration file passed
through the TFTP server.
Thanks,
Dan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 03, 2003
Looks as though there may be a problem with RSA key authentication and
switch for some reason... Does it behave different with either no
authentication, or password authentication?
Mark
On Thu, 3 Jul 2003, Anton Yurchenko wrote:
hello,
I have a test setup with 2 asterisk servers, each
Mark Spencer wrote:
Looks as though there may be a problem with RSA key authentication and
switch for some reason... Does it behave different with either no
authentication, or password authentication?
Mark
you are right, it is a rsa authentication problem. with other
authentications it work
That means that asterisk is sending SIP messages but gets no response from
the device.
Martin
On Thu, 3 Jul 2003 [EMAIL PROTECTED] wrote:
Hello All!
There is description of my problem with Asteriks below.
Asteriks CLI says:
File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded
Hi,
When I started printed out and read the two handbooks and the whitepaper which can be
found on the www.digum.com site on the Docmentation page..
I then installed Asterisk and started playing.. anytime I got stuck I asked the
questions on the mailing list and most times got the help I
Is there a way in Asterisk configuration to force the
use of specific codecs only... for example:
Never use GSM
Try G.723 if available with end point
Try G.729 if available with end point
Try G.711 if available with end point
Something like that?
Thanks!
Chip
__
When a connection is carried between two IP end
points, does the Asterisk server incur CPU usage to
pass the voice bearing circuit between the two end
points? Is it possible to have Asterisk setup the call
but hand off the voice traffic to be handled directly
between the two end points?
Thanks!
Here are a list of howto's. There are not found on the Asterisk site.
You will find most of your help in the mailing list archive.
http://asterisk.gnuinter.net/ - this howto has links to other Howto's
http://asterisk.drunkcoder.com/ http://asterisk.drunkcoder.com/apps.html
I do it this way:
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Background,adi-greeting
exten = s,6,Background,adi-main_menu
Then have the timeout extension as follows:
exten = #,1,Playback,adi-thanks
exten = #,2,Hangup
Thank you for your responce
I have also Pingtel SIP phone
I have encounter with the same problem on it. (Disconnect after some reinvite
messages)
But that I put in the Pingtel configuration (SIP_SESSION_REINVITE_TIMER) to
1200 and I have no disconnection after it at all.
So it seems that the
Message: 1
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie question
Date: Thu, 3 Jul 2003 13:14:44 +0300
Reply-To: [EMAIL PROTECTED]
Hi,
I have the same error only whan I start Asterisk with asterisk -vc
If I start it with safe_asterisk and
IIRC asterisk by default will not participate in the call between two SIP phones.. It
will help establish the session to the correct UA and then have nothing more to do
with it unless the call is transferred to another UA in which case Asrerisk will again
be involved in setting up the call..
this is NOT a '0' issue, it's a SAME (back to back) digit issue.
any set 00,11,22,33,44,55,66,77,88,99
and it's not xlite fault, that end is sending it, the * rtp end
is what's slicing.
Eris Riswanto wrote:
Hello guys,
Have you guys experienced problem with digit 0 from x-lite.
I cannot
a stupid question...
have you started the key from the private keys by issuing
init keys on the console ?
Matteo.
Il gio, 2003-07-03 alle 14:46, Anton Yurchenko ha scritto:
Mark Spencer wrote:
Looks as though there may be a problem with RSA key authentication and
switch for some reason...
Hi Frank-
I've had a similar experience with Asterisk over my first week, but have had
some recent breakthroughs. Asterisk/Digium will in the end save me a huge
amount of money over a typical Dialogic/high-level voice language GUI
approach, so it will all be worth it (I hope!)
But it has been
Title: Message
Hi,
I am new to Asterisk
and was wondering if Asterisk has the ability to act as a protocol
converter.
I have an H.323
network and I want to know if Asterisk can convert the signaling to SIP so I can
send it to SIP Addresses?
Thanks for your
help!
regards,
Sam
Michelson
On Thu, Jul 03, 2003 at 03:57:09PM +, WipeOut . wrote:
IIRC asterisk by default will not participate in the call between two SIP phones..
It will help establish the session to the correct UA and then have nothing more to
do with it unless the call is transferred to another UA in which
I need a recommendation on a good motherboard/processor
combination. I would like a
motherboard that has lots of PCI slots and works well with Asterisk without
problems getting drivers working, etc.
Onboard LAN would be nice to keep from using a slot. Plan to use RedHat
8 for the OS.
from my little experiences, use mb with less onboard stuff
as possible...
video+lan is ok.
unfortunately all have usb, try to keep usb controllers
minimum as possible.
I've used asus mb with ide+sata+audio+6usb+video+eth
and were pretty unstable. (even if devices are
disabled from bios)
With
Which codec will be tried first?
On Thu, 2003-07-03 at 11:02, WipeOut . wrote:
in sip.conf...
disallow=all
allow=g723
allow=g729
etc..
or
allow=all
disallow=gsm
etc..
Is there a way in Asterisk configuration to force the
use of specific codecs only... for example:
If I understand correctly, each codec has a cost associated with
converting it to another codec. I would think that Asterisk would
choose the lowest-cost codec first, but I'm not positive that's what
happens.
Jared Smith
On Thu, 2003-07-03 at 11:35, Eric Wieling wrote:
Which codec will be
On Thu, 2003-07-03 at 12:35, Brancaleoni Matteo wrote:
from my little experiences, use mb with less onboard stuff
as possible...
video+lan is ok.
unfortunately all have usb, try to keep usb controllers
minimum as possible.
I've used asus mb with ide+sata+audio+6usb+video+eth
and were
That I am not sure on but usually there is the ablility to choose the preferred codec
on the phone..
Which codec will be tried first?
On Thu, 2003-07-03 at 11:02, WipeOut . wrote:
in sip.conf...
disallow=all
allow=g723
allow=g729
etc..
or
allow=all
disallow=gsm
etc..
I have setup 2 Asterisk boxes both on gigabyte boards (one with onboard LAN and the
other without) and using Intel proc's (A P4 and a P3)..
So far I have had no problems..
Many people have had good results from AMD proc's as well but as I recall you must not
use the AMD optimised kernel and
Hi
please excuse if this seems obvious, but I am new to this and the SIP
section in the Asterisk handbook do not give any clues nor do the SIP
examples in there seem to represent real-world situations.
I am using Nikotel as a VoIP provider (for now) and I would like to
configure Asterisk to
How to turn off reINVITE messages on ATA-186?
Thanks in advance
--
Best regards
Vlad
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I've just started trying to use this functionality and I get the invalid
conference number message. Any ideas?
I started out with:
exten = 7315,1,Meetme,1234
and
confno = 1234
and then tried:
exten = 7315,1,Meetme
and
confno = 1234
and enter 1234 at prompt.
All give the same message.
Make sure you have a new line(blank empty space) at the bottom of your .conf files..
This can cause things not to work..
I've just started trying to use this functionality and I get the invalid
conference number message. Any ideas?
I started out with:
exten = 7315,1,Meetme,1234
and
yes, without a problem :)
Sam Michelson wrote:
Hi,
I am new to Asterisk and was wondering if Asterisk has the ability to
act as a protocol converter.
I have an H.323 network and I want to know if Asterisk can convert the
signaling to SIP so I can send it to SIP Addresses?
Thanks for your
Hello,
It is my understanding that on the softphone side, asterisk is only
responsible for establishing the session between two phones. If this is the
case, does it matter what type of audio codecs the two phones are using? And
if it does matter, are there any codecs that cause problems
Hi,
I have one, but all I can do for the moment is to register with ICH and then
call a SIP address from my Asterisk server.
BR,
Dan
- Original Message -
From: Humberto Atristain V. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 22, 2003 6:14 PM
Subject: [Asterisk-Users]
We currently have a Merlin Legend system. The voicemail is falling apart
(with the transition to a 10 digit timestamp on Sept. 8, 2001, the system
locked up and refused to take calls; the official solution is to change
the system time back to a year with a matching calendar). We are in the
Hi Lubomir,
Thanks for your answer.
My company needs SIP/H.323 conversion. As I understand it, the
conversion is only to the signalling not to the media itself.
Is it possible that your company can provide this protocol conversion
as a service to us that we can pay you for? This would help us
At 04:11 PM 7/3/2003 -0400, you wrote:
We currently have a Merlin Legend system. The voicemail is falling apart
(with the transition to a 10 digit timestamp on Sept. 8, 2001, the system
locked up and refused to take calls; the official solution is to change
the system time back to a year with a
On Thu, 2003-07-03 at 15:11, Steve Creel wrote:
We currently have a Merlin Legend system. The voicemail is falling apart
(with the transition to a 10 digit timestamp on Sept. 8, 2001, the system
locked up and refused to take calls; the official solution is to change
the system time back to a
On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote:
Hello,
It is my understanding that on the softphone side, asterisk is only
responsible for establishing the session between two phones. If this is the
case, does it matter what type of audio codecs the two phones are using? And
If the voicemail interface really is analog, then the identification is done
through DTMF codes. You'll just have to figure out the format of these
digits and let Asterisk deal with them as you want.
Most systems with analog voicemail ports work just like this. We've
interfaced Asterisk with a
On 3 Jul 2003, Steven Critchfield wrote:
On Thu, 2003-07-03 at 15:11, Steve Creel wrote:
Right now, the voicemail system (and auto-attendant) are connected to the
switch by 4 analog lines. Logic says that these are FXS cards in the
switch, like any other extension. The switch handles an
use r instead of m
You'll hear ringing , better than nothing.
Don't remember if moh works when dialling from sip phones...
Matteo.
Il gio, 2003-07-03 alle 23:11, Jim Archer ha scritto:
Hi All...
When I dial into my Asterisk box and then dial an extension, I here silence
until the person
I am using a (don't laugh) eMachines T2240 with a 3Ware Escalade RAID
controller and 384MB RAM, 3 80GB 7200rpm IDE drives on a RAID 5.
Installed extra chassis fan for $10.00. Machine has been problem free
for 2 months. I'm typing this on it now. Mandrake 9.1 loaded all drivers
flawlessly.
Ok, a little patch that adds a little functionality to call parking.
With that, you can pickup the older parked call, if many are in the
parking lot. The default exten to do that is 750, but can be changed
by setting parkpick = exten on parking.conf , like
[general]
parkext = 800 ;
Thank you for your help Steven.
Message: 8
Subject: Re: [Asterisk-Users] Drops due to codecs?
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: 03 Jul 2003 15:26:45 -0500
Reply-To: [EMAIL PROTECTED]
On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote:
Hello,
It
I just received my budgetone 100and the
difference between using sjphone and being at the mercy of my cheap Muse XL PCI
sound card is like comparing night and day. Now that I have some real hardware I
have my budgetone talking ulaw to my linux box then out to TNN (The Nufone
Network) for
The * code is not written yet. The Digium's cards rock... (ps I also
have a linejack in my drawer)
Dave
[EMAIL PROTECTED] 7/3/2003 2:10:23 AM
What do you mean a feature that is not present? I can dial out with
other apps...
-Z
- Original Message -
From: Andres Tello Abrego [EMAIL
Did you not know you could dial the parked number and pick it up
directly?
On Thu, 2003-07-03 at 16:36, Brancaleoni Matteo wrote:
Ok, a little patch that adds a little functionality to call parking.
With that, you can pickup the older parked call, if many are in the
parking lot. The default
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