- Original Message -
From: Kim Lux [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 26, 2005 8:13 AM
Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback
I updated to firmware version
Ahh, here we are...got a little more detail:
Linux ga1 2.4.21-27.ELsmp #1 SMP Wed Dec 1 21:53:57 EST 2004 x86_64 x86_64
x86_64 GNU/Linux
Gabe
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, 26 Jan 2005, Tobias Jönsson wrote:
On Tue, 25 Jan 2005, Peter Svensson wrote:
On Tue, 25 Jan 2005, Tobias Jönsson wrote:
No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier
1.0 releases too. Busy() may play a busy tone to the caller instead of
signalling busy
On Wed, 2005-01-26 at 00:07 -0800, Gabriel Afana wrote:
Ahh, here we are...got a little more detail:
Linux ga1 2.4.21-27.ELsmp #1 SMP Wed Dec 1 21:53:57 EST 2004 x86_64 x86_64
x86_64 GNU/Linux
Dump that crap. Use a normal, vanilla kernel so you can avoid RH
specific patches that are causing
As long as the bootloader exists on both disks, and boot order are including
both disks, there aren't any problems even booting with a failured disk.
But since SATA is (often) Hot Plug, you could change the failed disk while
running.
- Original Message -
From: Mark Eissler [EMAIL
Stewart == Stewart Nelson [EMAIL PROTECTED] writes:
Stewart If this is not available, I would be willing to put some
Stewart effort into enhancing the * MGCP stack, to also speak the
Stewart slave side of the protocol. Are there other Free users that
Stewart would be interested in contributing?
dhh == dhickman [EMAIL PROTECTED] writes:
dhh When I make a call, bellster anounces that I have no credits and
dhh says goodbye, but it still routes the call.
I just noticed another interesting problem: I checked that using
Congestion I can appropriately reject an incoming bellster call and
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 26, 2005 12:14 AM
Subject: Re: [Asterisk-Users] size and quality of
audioclipseffecttheplayback??
On Wed,
Hello All,
I have got my TE110P working at the hardware level, turned out to be a
dodgy cable causing the Yellow errors in zttool.
However, now I am getting yellow errors in asterisk, but zttool shows
nothing out of the ordinary.
here is my current config, and some asterisk console errors, any
Surely no other route would be tried in this instance, for as far as
all devices are concerned the A party and B party were connected
correctly, albeit in this instance to an announcement shelf device.
I agree that the A party has a right to be annoyed at the loss of
credit, but this has been
OK
I found it in modules.conf. looks like this:
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload = chan_alsa.so
;noload = chan_oss.so
Is this correct?
cheers,
Mick
Jim Kou [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
I trying to set up an h323 channel over TCP/IP network
to connect two
PBX.
I just read http://www.voip-info.org/wiki-Asterisk+config+h323.conf
but, it don't solve my dubs.
How could I use a h323 channel with asterisk?
Could anyone paste a part of h323.conf file? I am no sure how to setting
up
That's correct.
Make sure that the chan_oss.so work properly, if so you can use Dial
app. now.
Mick Hastings on 2005/1/26 04:50 wrote:
OK
I found it in modules.conf. looks like this:
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload =
Hello everybody, i got and asterisk and a CCM configured thru SIP, and in
the sip show peers appear
Name/usernameHostDyn Nat ACL Mask Port Status
CCM 10.60.27.138255.255.255.255 5060 OK
(1 ms)
but when i enabled sip debug in the
I can recommend you to use the chan_oh323 from inAccess Networks -
according to our experience it's much stable and bug free channel.
http://www.inaccessnetworks.com/projects/asterisk-oh323
Lubo
-
AppRadius Project: Full RADIUS AAA support for Asterisk PBX
Hi all
I have asked this question before but have not got any
helping input.
Im really new to this and need some explanation about
ASTCC.
So here is the question again.
In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards.
As I understand
Hi,
I'm trying to configure a Polycom IP Phone SoundPoint
500 to connect it to my Asterisk PBX but with no
success.
First of all, I downloaded the SoundPoint IP SIP
Administration guide I found on internet and then I
tried to make a boot server creating an FTP account on
my Mandrake 9.1 Linux box
On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote:
If you have it, can I get a copy please, or possibly can you send it to the
keeper of http://www.freedomphones.net/polycom/files/
I am looking for the latest boot image too.
1) I have the 1.4.1 firmware. To whom should I send
On Tue, 2005-01-25 at 15:55 -0800, Stewart Nelson wrote:
If this is not available, I would be willing to put some effort
into enhancing the * MGCP stack, to also speak the slave side of
the protocol. Are there other Free users that would be interested
in contributing?
As another */Free
Hello all,
I am planning to connect my Asterisk with the FWD and/or iaxtel
networks.
Two mounths ago, I just used the iaxtel network, and i remember I have
trouble with this network, I can not place a call. The service do not
wotk.
With FWD I alwais can place a call, I never get an error from
Hi.
A friend of mine came asking about this VOIP provider.
I havent heard of them so I thought I might ask the
list.
Anyone has any experience using them (http://www.optonline.net/Home, http://www.optimumvoice.com/index.jhtml)?
Not just with Asterisk, even with their supplied
Sir/Mam,
Good PM to all! I'm new to asterisk but I was able to setup a asterisk server using softphones.
I have some questions in mind, I have a working asterisk server and I want to add digium cards w/ a telephone line. Will it be able to forward a call from the a person who is in the U.S. using
hi
could I have a look at this?
I really need it, urgently, so please..
roy
On Jan 20, 2005, at 12:17, Ben Merrills wrote:
I've not released the source yet, I asked last week on the mailing
list for people to send me over some example queue_logs, because so
far I've only been able to test
Hello All,
Im on the technology committee for a fraternity at the University of
Illinois.
Were looking into moving from our current party line (one line shared
between every two rooms) system to a PBX with voicemail in an effort to lower
our monthly phone bill and provide better
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
It also means that you need a permit from the Israeli ministry of
communications cause you're acting as an international call provider.
Can't be done here.
Being a CableVision customer I get harrassing phone calls from these
guys all the time trying to sell me their OV service.
Firstly it's closed. They won't allow you to bring anything to their
network. Secondly it uses G729 so there's no faxing etc (although you
can buy that for extra cost).
Well, it sounds to me like his phone actually IS sending you keypresses.
You stated that it goes silent on his end while you are hearing his DTMF
tones. Sounds like the phone is silencing his end, as it would if he were
intentionally dialing out. I am guessing that he has a bad phone or
something
Ismael Gil wrote:
I am planning to connect my Asterisk with the FWD and/or iaxtel
networks.
There is nothing stopping you from connecting to both simutaniously, in
general there's only small amount of overhead to remain connected to a
foreign network.
Although there is no overhead from listing
- Original Message -
From: K J [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 10:06 PM
Subject: [Asterisk-Users] Tie web application to VOIP
I want to tie my web application (built using .NET + MS SQL Server)
into a VOIP service so that users
Hello,
I just checked out the latest CVS and compiled and now
get the following error:
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
Jan 26 13:03:51 WARNING[27081]: config_old.c:27
ast_load: ast_load is deprecated, use ast_config_load
instead!
== Parsing
Hi List
Just a little bit OT, but then again perhaps an information that could be of
great value for a lot of administrators !!
Does anyone have experience with how to setup VoIP QoS for outgoing data
through a Cisco PIX (515) ?
I believe that it should be possible to give higher priority to
Asterisk is software installed on linux installed in a PC with a hard
drive.
When I say it might not come up after a power failure I don't mean
Asterisk, I mean Linux.
The hard drive might fail and you can kiss you system good bye.
Legacy PBXs don't have that problem. The configuration there is
Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in
known good telco lines in various combinations on channel 1 through 4 -
problem is channel 1, not anything external. So after seeing lots of
stuff on the list re: TDM400's I power cycled, removed board and let
linux
OK.
You're wearing me out.
IF linux boots Asterisk can surely load automatically.
What if linux DOES NOT boot after a power failure?
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 25, 2005 10:52 PM
To: Asterisk Users Mailing List -
I have installed asterisk from the CVS source on Jan 7th and I am having
problems getting call transfers working.
features.conf contains:-
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1 ; One Touch Record
atxfer = *2
:)
Get an APC power switch, hook it up to a network in the vicinity of your *
box and make sure you can reach it from the outside. If the box fails to
boot you can remotely power cycle it.
If you need a rocksolid solution have a look at astlinux that can boot *
from a compact flash card in
hi all,
Im trying to configure a * server with FXO and FXS cards. Basically
what i want to do is be able to recieve calls from PSTN and dailout as
well...but im really very confused with how to handle the 30 channels coming
in on the PRIwhat would be the best hardware to use n stuff..
Hi
I'm confused by the asterisk WIKI syntax for DISA. I want to only let a CID of
say 1234567 pass through DISA, which calls an extension of 333
In reading the documentation, I thought it should look like this
exten = 333/1234567,1,Authenticate(1234567)
exten =
If you need a rocksolid solution have a look at astlinux that can boot *
from a compact flash card in read only mode which makes it very hard to
break :)
You should be able to boot Asterisk using slackware as a base from a 64M CF
card or even from a 64M bootable USB memory key. If you use
David == David John Walsh [EMAIL PROTECTED] writes:
David I agree that the A party has a right to be annoyed at the loss
David of credit, but this has been tradition within telco's for as
David long as i can remember, as a call channel costs significantly
David more bandwidth than signaling
I
Hello,
FXO and FXS cards are only for analogic lines, but if you need connect *
with a PRI, maibe need añother kind of hardware.
See the digium or AVM homepages. There you'll find what are you looking
for.
Ismael.
On Wed, 2005-01-26 at 17:31 +0500, Hussain Umair wrote:
hi all,
Im trying
Well, in our country (dont know others) we have different plans for
residential users, other plans for commercial users, about 8
international long distance phone services that you can select at
dialing time and 3 carriers for domestic long distance.Ohh, and our
cellphone providers (TDMA/CDMA/GSM)
Did you try to boot without lan just the power ...
I've had this same problem to and rebooted the device without lan
connection
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Louis-David Mitterrand
Sent: woensdag 26 januari 2005 11:36
To: 'Asterisk
If you have it, can I get a copy please, or possibly can you send it to the
keeper of http://www.freedomphones.net/polycom/files/
I am looking for the latest boot image too.
1) I have the 1.4.1 firmware. To whom should I send the files? There is
no contact info in this web site.
2)
On Tue, 2005-01-25 at 20:57 +0100, Wilson Pickett wrote:
L( ) option is
applicable here? And, if your version of Asterisk doesn't have a Dial app
with the L( ) option,
will it be worth your while to upgrade to have the L( ) option?
A third question might be in what version was this
Off topic but I am after a DECT phone to connect to my sipura 3000 that has
a FSK VMWI light or flashing envelope on the LCD screen. Any ideas
Chris
___
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Asterisk-Users@lists.digium.com
Here is what I am attempting to do (which works well on Cisco Call
Manger). I have some 7960's that have multiple lines on them. The
second line specifically is a helpdesk line that is shared among
multiple phones. Here is how I am making that line ring on multiple
phones, maybe you have
All,
One of our customers is using a Telrad PBX, we are providing
phone server through asterisk via a T1 using em directly connected to
the Telrad system. We're using a T1 cross cable as normal, the T1 part
works great. No alarms. When we try and dial out the Telrad using a
direct trunk
On 26 Jan 2005, at 13:11, Chris Stenton wrote:
Off topic but I am after a DECT phone to connect to my sipura 3000
that has a FSK VMWI light or flashing envelope on the LCD screen. Any
ideas
Can you post the relevant extension from sip.conf and the contents of
voicemail.conf.
Also, check
Here is what I am attempting to do (which works well on Cisco Call
Manger). I have some 7960's that have multiple lines on them. The
second line specifically is a helpdesk line that is shared among
multiple phones. Here is how I am making that line ring on multiple
phones, maybe you
Im trying to install [EMAIL PROTECTED], Ive just
downloaded the latest cd from soundforge. I can get it to install ok (network
card didnt auto configure but I worked out how to use netconfig).
I worked out how to add a few grandstream budgetone fine. Worked
out how to upload music etc.
Ivan Meic (Vox Mundi) wrote:
Stuart,
Can you plase specify in which mode are you using your
hfc cards ? You said ptp, but are they working as NT or TE ?
Ivan,
I'm using the hfc cards in ptp mode connected to the pstn in TE-mode.
During testing we used the same setup on a different isdn-line in
Hi
I'm trying to deploy asterisk, but I can't seem to find documentation for
the TFTP server to run the cisco 7960 ip phones'
I was told before that you need it and it could run on linux.
Thank You
,jm
___
Asterisk-Users mailing list
Hello!
Doesn't matter which TFTP server you will setup. Any kind of TFTP will
do it (Linux, Windows, Solaris, FreeBSD...).
BR,
Alen
Jose Cruz (Branders IT) wrote:
Hi
I'm trying to deploy asterisk, but I can't seem to find documentation for
the TFTP server to run the cisco 7960 ip phones'
I was
Thanks
But how about the config files (SIP...) that needs to be inside the tftp
server, where can I get a sample of that?
That's where the images for the firmwares of the ip phones come from, on
boot right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
no audio with echosquelch=0 in capi.conf
can someone compile chan_capi changing the Makefile with
CAPI_ES disabled
and
CAPI_GAIN enabled
no audio in the channel
I had to disable the CAPI_ES and CAPI_GAIN to get it working
Can someone confirm this?
Sergio
dhh When I make a call, bellster anounces that I have no credits and
dhh says goodbye, but it still routes the call.
Even if you have no credits, we'll try to route the call via DUNDi.
___
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Asterisk-Users@lists.digium.com
I'm not sure if this will work with your cisco's but I can guarantee
that it works with the grandstreams.
This is what I use to update my 4 phones, running on my main winxp
machine and it's free for non commercial use.
http://www.weird-solutions.com/product/tftpc2000.html
Cheers,
Dean
-Original Message-
From: Chris Stenton [mailto:[EMAIL PROTECTED]
Sent: 26 January 2005 13:12
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] off topic - DECT phones with FSK
VMWI in the UK
Off topic but I am after a DECT phone to connect to my sipura
3000 that
Yes, this is frustrating I know. In fact the wiki could be updated to
provide this info. Basically if you have the phones out of the box
(brand spankin new) then you probly have the SCCP image installed on it
by default. Your tftp server root will need a number of files to start
if this is the
Sorry my mistake, wrong link, here is the correct one.
http://www.weird-solutions.com/product/tftp-desktop.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Wednesday, January 26, 2005 9:13 AM
To: Asterisk Users Mailing List -
Jose Cruz (Branders IT) wrote:
But how about the config files (SIP...) that needs to be inside the tftp
server, where can I get a sample of that?
That's where the images for the firmwares of the ip phones come from, on
boot right?
Jose,
Under Mandrake, to install the tftp program is, urpmi
Hello everybody,
I'm having little trouble (well, pretty big trouble) with HFC-S card and
Asterisk. My idea is to do VoIP/IAX link between two HW PBXen using two
Asterisk PC boxen with ISDN cards in them. AFAIK HFC-S cards must be in NT
mode for this installation, they must behave like state
That's very interesting, because we do the exact same thing and all the
phones light up (with line mailbox flashing).. What SIP ver are you
using on the 7960's? However it sounds like 135 isn't registered on all
the phones? What we did is bind the lines to multiple phones, 203 (our
tech mailbox)
you'll have to get them off Cisco's site, there was a posting recently
stating where to obtain these without a cisco login.
Anyone care to remind us of where this was??? (did a quick search and didn't
see anything immediately).
Since we are outside the US, Cisco US refuse to 'sell' us a login
I am trying to use the _ALERT_INFO variable with ParkAndAnnounce. My idea
was to have the phone, a Polycom IP500 auto answer so you could hear the
annoucement of the parked extension over the speaker. This variable works
fine with the normal Dial application, but seems to be ignored by
Get a TFTP for Linux if you use Red Hat or Fedora Core get the server
there : http://dag.wieers.com/packages/tftp/
it's also available on the install CD...
Then to know what file you need to change your Cisco 7960 phone from
Skinny to SIP go to this website as it explain how to do it. If it
Got fed up going round in circles in the end. all for $8 worth of
access :(
Technically, Cisco wants you to pay for those images :)
___
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Asterisk-Users@lists.digium.com
Rich Adamson wrote:
Here is what I am attempting to do (which works well on Cisco Call
Manger). I have some 7960's that have multiple lines on them. The
second line specifically is a helpdesk line that is shared among
multiple phones. Here is how I am making that line ring on multiple
Got fed up going round in circles in the end. all for $8 worth of
access :(
Technically, Cisco wants you to pay for those images :)
Indeed, and I would if Cisco made it Technically possible! :)
___
Asterisk-Users mailing list
I'm trying to deploy asterisk, but I can't seem to find documentation for
the TFTP server to run the cisco 7960 ip phones'
I was told before that you need it and it could run on linux.
Try 'man tftpd' at your favorite unix command line.
___
Martin
What website? I think you forgot to put the link to the site on how to do
it.
Thanks
,jm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy
Sent: Wednesday, January 26, 2005 6:41 AM
To: asterisk-users@lists.digium.com
Subject:
Can't you just create a different context for inbound and outbound
calls? Then specify your codec preference order in there. I don't think
you can specify the bandwidth= parameter within a context other than
the global one though.
-mark
On Jan 25, 2005, at 6:13 PM, [EMAIL PROTECTED] wrote:
I
I am getting to my wits end with these phones (and so is my boss). I am
getting an random echo on these phones and I have an issue opened with
Polycom and its been in their research and development department for
almost a month with no results.
I have noticed that I get a message RFC3389
I run an automated information retrieval system, using Asterisk. Fairly
often the system misses a dialed digit. Our codes are all 4 digits, see
lots of logs with:
4199 - OK
530 - Invalid code
330 - Invalid code
5330 - OK
As callers experience skipped codes. We're
I'm having the same problem with Voicepulse connect using IAX2. So, no,
it's not better IMHO. And I've been thinking about switching to SIP to
see if the problem goes away (I'm very reluctant to do so though) but
it's hard to know if the problem lies with Voicepulse (or Broadvoice in
your
When I access the Directory() and use it to call an extension, the
origination hears garbled or inconsistent ringing. The termination
side rings normally and the conversation is clean in both directions.
Kurt
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Asterisk-Users mailing list
It would be really nice to see whatever patches they develop for
Asterisk or at least get some hint of where the problem lies.
-mark
On Jan 25, 2005, at 9:41 PM, Brian Dingman wrote:
LiveVoip has a problem with Asterisk users on versions less than 1.0.3
If
you are not using that version
Is it not possible to use sip debug or ethereal to see what digits
arrive at your site? (or do you already know the digits are mutilated
before getting to you?)
I'm having the same problem with Voicepulse connect using IAX2. So, no,
it's not better IMHO. And I've been thinking about switching
My phones were running firmware version x.18.
There was a field that allowed me to select automatic updates and how
often. I selected yes and set it to 1 day. (I thought maybe 0 days
would cause it to update immediately, but all it caused was an error.)
There was a field for http updates. I
Just to let everyone using [EMAIL PROTECTED] know that my livevoip DID now works
without any changes to asterisk!
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Eissler
Sent: Wednesday, January 26, 2005 10:00 AM
To: Asterisk Users Mailing List -
-Original Message-
From: Michael 'Moose' Dinn [mailto:[EMAIL PROTECTED]
You should be able to boot Asterisk using slackware as a base
from a 64M CF
card or even from a 64M bootable USB memory key. If you use
ReiserFS or
something similar for the drive that stores all your
I use it for my 7960's at the house and it works fine.
dean collins wrote:
I'm not sure if this will work with your cisco's but I can guarantee
that it works with the grandstreams.
This is what I use to update my 4 phones, running on my main winxp
machine and it's free for non commercial use.
I am running RedHat Fedora Core 3 with their kernel-2.6.10-1.741_FC3 kernel,
and if I use nethdlc in zaptel.conf, when I run ztcfg I get the message:
ZT_CHANCONFIG failed on channel 11: Function not implemented (38)
Does anyone have ideas on how I can work around this issue?
T1 channels 1-10
Oops forgot to give the second link... here it is :
http://www.wheely-bin.co.uk/cisco/
For people requesting SIP images for Cisco phones there's a few link
where you can get them here's 2 I found :
http://www.m4tr1xx.de/cisco/ (there's SIP 6.3, 7.0 and 7.1 there)
On Wednesday 26 January 2005 14:34, Paul Brock wrote:
you'll have to get them off Cisco's site, there was a posting recently
stating where to obtain these without a cisco login.
Anyone care to remind us of where this was??? (did a quick search and
didn't see anything immediately).
Since we
I have my asterisk box showing an incoming call in the logs. But I get a
message on the phone that the number is busy and I have to leave a message.
I found out that the message was on my broadvoice voicemail.
This happens everytime.
I also saw something in the logs that says congestion for
Are you kidding? DTMF handling in Asterisk is the worst, it might not just
be Asterisk. Dropped digits is a common thing. INBAND basically plays the
sound of the DTMF digit, that's why G711 is the only codec it'll work on,
it's the only high enough quality codec to play a touchtone and it be
I believe the current implementation for vm notification is to use
a sip 'notify' message to turn on the mwi, and the sip protocol
implementation within * does not support sending 'notify' messages
to multiple phones. (E.g., how would * even know how many phones
you are trying to ring via the
LiveVoIP did not issue any end user patches last night. They had a
problem connecting to Level 3's network. LiveVoIP claimed the problem
was with asterisk users, I have not upgrade or install any patches and
all is fine now.
My main problem with LiveVoIP has been the LACK of customer service.
Edgar de Leon wrote:
Hello everybody, i got and asterisk and a CCM configured thru SIP, and in
the sip show peers appear
Name/usernameHostDyn Nat ACL Mask Port Status
CCM 10.60.27.138255.255.255.255 5060 OK
(1 ms)
Just a guess,
Hi,
Is there a way to restart the DISA to the enter phone number? For
instance, Bell Calling Cards let you hit # at any point which lets you
enter another number to call. This is useful to reduce the number of
digits dialed and to utilize per-minute calls.
I was not able to find anything on the
I have two Vonage lines and one Lingo line. I would like to drop them
and go with BroadVoice. I would need to have three to four lines from
BV. I should be able to configure Asterisk to handle all the SIP
connections? Right
Thanks, David
PS its a home PBX.
On Tue, 2005-01-25 at 11:39, Jay
However, the second route tried by bellster ended up with
This is 9:25 local time, calls are only permitted from ... to
It means that the remote asterisk accepted the call to play the
message, instead of using Congestion to use another route or fail.
The playback command has a no answer
Bill Lattner wrote:
Right now our setup is looking as follows:
12 ch T1 with 60 or 80 DIDs (using Digium T100P)
P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk)
41 Sipura ATAs (SPA-1001)
41 Sipura ATA's? Wow. that's a lot.. Have you thought about using
channel banks or the
I used them for a while and it was great. The service went down twice
(once for an unknown reason for about 30 minutes and once on Christmas
due to call volume).
Call quality was excellent and they provided CNAM. It was all I needed.
On Wed, 2005-01-26 at 13:03 +0200, Shoval Tomer wrote:
Hi.
However, the second route tried by bellster ended up with
This is 9:25 local time, calls are only permitted from ... to
It means that the remote asterisk accepted the call to play the
message, instead of using Congestion to use another route or fail. I
lost one credit without having the
I have a problem, when in a call with a grandstream phone it disconects
around 4 mins into a conversation, I have tryed it with different
providers, and still get the same results. But is wierd, in the
grandstream phone i get a fast busy, while on the other side the call is
still stablished AND
Adi Linden wrote:
I believe the current implementation for vm notification is to use
a sip 'notify' message to turn on the mwi, and the sip protocol
implementation within * does not support sending 'notify' messages
to multiple phones. (E.g., how would * even know how many phones
you are trying to
Hi,
Does someone know an ActiveX IAX softphone?
I need a free softphone to connect with Asterisk from a web page.
Regards
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