Scott Stingel wrote:
Some questions:
What country are you in?
Is there anything else connected to the line from the PSTN? It sounds
like you have a marginal condition, such as insufficient loop current
perhaps.
Do have any features, such as call waiting, on the line?
Do you know how far
Hi friends !
Can anybody help me to configure asterisk with ser so that I can share
the load of the asterisk with ser server. I have tried it but my
asterisk is not showing registrations of the user agent, as given in the
asterisk wiki/asterisk+at+large. I don't know what is the problem, but
Hello Deepak,
1. don't post multiple times. it's annoying. enough said.
2. run asterisk in verbose mode (start it with asterisk -vgc),
place a call from a SIP endpoint behind SER to the asterisk server,
and see what happens in the asterisk CLI.
3. if you don't see anything there, get ngrep
Paul Hales wrote:
It now works - but only in the latest (1.5+) firmware releases.
Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites.
Regards,
Richard
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Hi,
-Original Message-
How knows where I can get a Dutchphone number for asterisk?
Pilmo is not delivering one for home use.
I think you are physically outside the netherlands, right ? Would you care
for an 087 number ?
Florian
___
Steve Prior wrote:
I've got a Sipura SPA2000 ATA basically working (I can place calls
between the
extensions plugged into each of its ports) and part of that was
setting up the
dial plan on the SPA2000 to match the one in Asterisk. This seems
like a pain
to deal with long term and I don't know
soft hangup channel name
-Abhishek
Drishti-Soft Solutions Pvt Ltd
http://www.drishti-soft.com
On 5/2/05, Asterisk guy [EMAIL PROTECTED] wrote:
1 after giving command oh323 show channels,
i want to disconnect a call, is there any command to disconnect a call?
2 how asterisk kill a
Hi,
I am testing an AP200 from addpac i m trying to make it register with
Asterisk. It manages 3 protocols (sip, h323 and mgcp).
If i use sip, i keep getting this messages:
May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration
from 'sip:[EMAIL PROTECTED]' failed for
Hi
I've installed successfully:
- PWlib v1.6.7 library
-Openh323 v1.13.5 library
-asterisk-oh323 v0.6.5
and so the modules chan_oh323 is installed successfully
Now I try to install chan_h323
First question: is this necessary?
I edit the Makefile in the directory
On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and
All,
I seem to be confused :(
Meetme won't work with the message That is not a valid conference
number, please try again even with the simplest of configurations.
Having trawled the list archives, wiki and harrased people on
#asterisk I've come to a dead-end.
I compiled ztdummy last night from
Same as subject!! I want to build my own * box - and i would like some
recommendations for the FXO cards.
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On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote:
2) There isn't anything like what you want. I know, I want the same
thing. There is no phone out there that will do this with any protocol
that asterisk uses. This is the one major failing of asterisk ( and
voip in general. I smell
Why are there FXO cards, and FXS cards? What's the difference, and why
is it needed? Modem cards, seem to be able to dial out, and receive
calls, so why are these cards different?
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On Sat, 2005-04-30 at 15:25 +0200, Joris Vandalon wrote:
Hi,
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
Queues are not an options because as soon a call comes in to a queue
there is no
On the subject of phones..
The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model)
phones can all do what he wants. ie, have multiple lines with blinking
red lights when a call arrives on that line.
How about phones that can indicate if an extension is busy or not - eg,
Busy Lamp
On Sun, 2005-05-01 at 22:43 -0500, Eric Wieling aka ManxPower wrote:
Eric Wieling aka ManxPower wrote:
Daryll Strauss wrote:
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog
Hi,
Im trying to set up the following situation.
Ive got several inbound server en one server
where we can make an outbound call.
If a caller calling to a inbound asterisk server (say
serverA) then I want to connect the call to the outbound server(say serverB)
So an inbound caller
[EMAIL PROTECTED] a écrit :
Hi
I've installed successfully:
- PWlib v1.6.7 library
-Openh323 v1.13.5 library
-asterisk-oh323 v0.6.5
and so the modules chan_oh323 is installed successfully
Now I try to install chan_h323
First question: is this necessary?
No, it's or oh323 or h323. I suggest
Hi All!
I am using Asterisk Stable 1.0.6 . Now I want to add another application
like app_chanspy in it. I have downloaded its source file but how can I
merge this application along with my already running asterisk ? Any
comments suggestions are appreciated ...
Thankyou,
Usman.
hello,
newasterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone
Hi,
First, do forgive any syntax or language errors as English is not my mother
tongue.
To make a 'long' story short,
- let's look at a phone home.
The public switch (the very big machine at your telco) provides power to the
line. This power gives energy to your phone.
This is how a 'standard'
I have a TDM400P I am trying to install but I need a power connector
extender to be able to get power into the card.
In the meantime can the card run without the power connector if it has only
one module on it?
The power connector is only needed if you have fxs modules installed.
The fxo
On Mon, May 2, 2005 1:12 am, Geoffrey Sachs said:
Thanks for the info.
What hard drives are you using ide or serial ata. Does it make a
difference. Thanks
There have been some references recently regarding disk drive types
relating to tdm400 noise problems.
Has anyone established there is a
Mr AG!! wrote:
Why are there FXO cards, and FXS cards? What's the difference, and why
is it needed? Modem cards, seem to be able to dial out, and receive
calls, so why are these cards different?
FXO card = plug to phone line
FXS card = plug to regular phone
FXS provides power to phone. FXO
Hello!
I would like to use meetme app for audio conference, but I have a problem with
the audio delay: it is increasing as time elapses. One way delay can reach 5
secs after 15 minutes usage.
I have read all the mails in the list archives, and the open case for the bug:
Any *cheaper* ones?
--Rob
You want a _cheap_ reception phone? I don't think you are going to get this.
Chris Mason
www.anguillaguide.com
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Hi
Folks,
I hope you can help
!!!
I have going
arround in my [EMAIL PROTECTED] 0.9 to have the
conference rooms and MoH working together. Everything works perfectly except
this two features.
I believe that all
has to do with the ztdummy interface.
After all installed
I ran 'yum -y
--- Tim Connolly [EMAIL PROTECTED] wrote:
Is NAT=yes on, are you behind a firewall? Give us
some connectivity details.
Usually when you see maximum retries, its because
you have one-way
communications with the far end for some reason.
Are
you setting externip
statically?
Hi,
FXS card = plug to regular phone
FXO card = plug to phone line
The trick I use is:
FXO with a 'O' as in Office.
This is where you plug your phone
A FXO card emulates a phone (receives power)
FXS with 'S' as in (public) Switch
This is the part that gives power
A FXS card emulates a
Yes, I am physically living in Rio de Janeiro, but I am going back to
Holland. I have at this moment a dutchphone connection with a 020 number.
I think for people to call me that will be cheaper and easier to accept then
a 087 number.
How every people who would like to call me can not belief that
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a bigger set of
phones.
Basically, this is easy,
I can do this in the
Thanks for the info.
What hard drives are you using ide or serial ata. Does it make a
difference. Thanks
There have been some references recently regarding disk drive types
relating to tdm400 noise problems.
Has anyone established there is a correlation between drive hardware
and
You want a _cheap_ reception phone? I don't think you are going to get
this.
Heh. I had a sneaking suspicion that was going to be the answer 8)
--Rob
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Mr AG!! wrote:
Why are there FXO cards, and FXS cards? What's the difference, and why
is it needed? Modem cards, seem to be able to dial out, and receive
calls, so why are these cards different?
For one thing, modem cards do not generate a ring voltage (they just
pass it thru from the telco.
Adam Goryachev wrote:
The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model)
phones can all do what he wants. ie, have multiple lines with blinking
red lights when a call arrives on that line.
The polycom ip600 and cisco 7960 both have 6 lines available.
Regards,
Adam
I am
Take a look at the Polycom 360 if you only nee 12 lines. otherwise look
at the Snom 220 with a sidecar (up to a total of 3 side cars may be
added for a total of 65 lines in the extreme need.)
Max W . Blackmer, Jr.
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If the problem is with libtiff, its a problem with every version i've
tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2)
On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote:
Me wrote:
Hi all,
I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript.
Hi,
does anyone know if x-lite supports the callto://name syntax on web
pages as skype does?
Kib
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td wrote:
-- Executing NoOp(Zap/4-1, ) in new stack
-- Executing Dial(Zap/4-1, SIP/tdhome) in new stack
-- Called tdhome
Same problem here. Any ideas?
Deti
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Here is an excellent document explaining the differences between FXO and
FXS.
http://www.google.com/url?sa=Ustart=5q=http://www.patton.com/technotes/fxs_fxo.pdfe=7385
Also you can look at Digium's site for their description, which
describes it from a stand point of Asterisk as the PBX.
In article [EMAIL PROTECTED],
Georg P. Israel [EMAIL PROTECTED] wrote:
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a
On Mon, May 2, 2005 8:24 am, Rich Adamson said:
To help identify the source of the delays, I built a new system this
weekend from scratch. When that is complete, I'll use it to compare
the differences in motherboards, OS distro's, and maybe kernel versions.
Very good Rich, the results of that
Hi friends !
Can anybody help me that how to use ser with asterisk server so that ser
can work like the front end of the asterisk and all other features of
the asterisk can be used.
I have tried the configuration given in asterisk-wiki/at+large but could
not succeed, still my asterisk in not
Send an example TIFF file, and I will investigate.
Regards,
Steve
Me wrote:
If the problem is with libtiff, its a problem with every version i've
tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2)
On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote:
Me wrote:
Hi all,
I'm trying to use spandsp
On Mon, May 2, 2005 9:01 am, Kim Culhan said:
Patches to the zaptel drivers are described on the Mantis link above.
El wrongo kimster, they're described in this post to the asterisk-bsd list:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000719.html
The patches are in this post:
Fix the date on your PC.
Deepak Dhiman wrote:
Hi friends !
Can anybody help me that how to use ser with asterisk server so that ser
can work like the front end of the asterisk and all other features of
the asterisk can be used.
I have tried the configuration given in asterisk-wiki/at+large but
Hi
I am using Phonejack PCI card connected to analog phone.
I've installed this card succesfully but i get no dial tone.
Have you suggestions?
Thanks Ale
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ide
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Geoffrey Sachs
|Sent: Lunes, 02 de Mayo de 2005 12:13 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
|
|Thanks for
Adam Goryachev wrote:
On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote:
2) There isn't anything like what you want. I know, I want the same
thing. There is no phone out there that will do this with any
protocol that asterisk uses. This is the one major failing of
asterisk ( and voip
Hi all.
Dopes someone know how I can move a key license of the g729
codec from one to another machine?
Find nothing usefull @ the wiki.
Thnx 4 help in advance.
Regards.
-Peter
--
Please no HTML, I'm not a browser
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The user name is the extension and the password is always the same. Not hard
to configure.
Yes, but each of those 6 lines on the 7960 must have
their own seperate SIP username/password. And if you are a
small office with 6 7960s, thats 36 username/passwords.
Trevor Peirce wrote:
Steve Prior wrote:
the SPA2000 does for me over the one in Asterisk. Is there a way to
disable
the use of the SPA2000 dialplan so I don't have to keep it in synch?
Or is
there some reason why it would be a bad idea for me to do so?
Sure just put x. as your dial plan and
Actually called Digium with this exact question last week. They said
that you can register the new license on the new server provided that
you ony registered it once before. They said there is no unregister
script to unregister the license from the old server, however. If you
have already used
Yes same provess you did to register the license in the first place.
You can rereg the license I think 3 times or so before you have to
call Digum and have them manually change what your license is tied to.
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- Original Message -
From: Tim Connolly
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Kerry Garrison'
Sent: Sunday, May 01, 2005 2:50 AM
Subject: RE: [Asterisk-Users] Broadvoice limits???
Broadvoice.
Seems to be no limit on inbound, but I found any channels after 5
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests, but
it is doing it now. I will try to track down the specific change tonight ...
My solution for now is to Answer() the call before dialing out. I changed all
of my
Hi.
I'ved registered it for 2 times, so I've got to contact digium.
Thnx 4 info.
On Mon, May 02, 2005 at 10:26:35AM -0400, Pedro wrote:
Actually called Digium with this exact question last week. They said
that you can register the new license on the new server provided that
you ony
Chris Mason (Lists) wrote:
The user name is the extension and the password is always the same. Not hard
to configure.
With the SNOM 220, you have five buttons/lamps that can be used as
line appearances--these buttons can each register to a different SIP URL.
Each sidecar has 20 buttons/lamps,
user monitor application
-Wix
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Hi all,
I try to dial via a Softswitch a number :
exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf)
And my sip.conf
[SIP-OUT]
type=peer
host=10.XX.XX.XX
defaultip=10.XX.XX.XX
disallow=all
allow=g729
allow=ulaw
allow=alaw
context=Ipnotic
canreinvite=yes
nat=yes
dtmfmode=rfc2833
But
Da: Aurelio Forese
Inviato: luned 2 maggio
2005 16.13
A:
'asterisk-users@lists.digium.com'
Oggetto: ExtensionState problems
using Asterisk API
Im trying to write a web application in php to
monitor the extension state of my asterisk peers. My application is working but
Here is how Digium license works.
1)You are allowed to register 2 times , for which digium license server
does not object.
2)If you want to register it a third time on a different server, send an
email to digium and they increment (or decrement the registrations, in
the real sense) the number, so
The cisco 7960 works well with * and SIP.
Out of curiosity I loaded the ccm version 7.1 and tested it briefly with
CVS HEAD * and latest chan_sccp.
The interface when using ccm load on the phone is certainly different.
Things I don't see how to fix are:
o Setting the date and time on the phone
On May 2, 2005 10:31 am, Charlie Watts wrote:
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests,
but it is doing it now. I will try to track down the specific change
tonight ...
My solution for now is to Answer() the
I could really use some input here, forgive the OT nature, but my
problem is related to asterisk and voIP on a DSL connection and
becoming a big mystery.
I noticed about three weeks ago a lot of UNREACHABLEs that became
REACHABLE 10 seconds later. After studying this a little, it happens
that the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Guy Decarpentrie wrote:
Hi all,
I try to dial via a Softswitch a number :
exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf)
And my sip.conf
[SIP-OUT]
type=peer
host=10.XX.XX.XX
defaultip=10.XX.XX.XX
disallow=all
allow=g729
Hi,
Can anyone think of a way to use asterisk as the voicemail system for
a Nortel phone system?
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Hello Everyone,
How can I control the time Asterisk reregisters with the IAX Provider.
The PPPoE ISP IP address sometimes address changes and the system
doesn't reregister and incoming calls are disabled.
Right now the only thing I'm able to do is Restart the server, that
seems to solve the
Is there a feature code you can dial after beginning an atxfer (*2)
that will bail out and return you to the caller. Let's say I want to
transfer to the CEO of the company, but only if he is available. Once I hit
*2, punch in his extension, I don't of anyway to cancel out. If I hit * or
Hi all.
Is there anyone who have a big experience with large scalable voip
setup and want to share some experience, knowlegde?
I need to handle a lot concurrent calls, to pstn and to sip gateways'
The current setup can't handle the load anymore.
I've some solutions in mind, but don't know if it
Simon Morris wrote:
On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or
On May 2, 2005 11:07 am, Matt wrote:
Can anyone think of a way to use asterisk as the voicemail system for
a Nortel phone system?
use a couple ATAs or an 8 port ATA card and wire them up to FXO ports on *,
have the extensions callforward-busy/unavail to the analog extensions. * can
take the
On May 2, 2005 10:56 am, Wilson Pickett wrote:
The phone company here has, after being evasive aboput checking the
DSLAM, claimed they did everything possible, changed our DSLAM
connection, tried every piece of equipment on their end. Ditto the ISP
who has been very cooperative.
Can you get
Hello
I've installed asterisk with [EMAIL PROTECTED] package with h323 support.
I've a Digium TDM10B card and we have a quintum voip gateway. I'm
trying to make call with an analog phone plugged to that card through
our quintum with h323 protocol.
How to confgure related files? Any help welcome.
Hi all,
I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I
don't need incoming calls (no need DIDs). Could someone tell me who is the
best IAX service provider for me? I want unlimited monthly basis or yearly
basis service. my DSL is 128kbps.
Thank You
Kumara
On Monday 02 May 2005 16:07, Dan Levine wrote:
Hello Everyone,
How can I control the time Asterisk reregisters with the IAX Provider.
The PPPoE ISP IP address sometimes address changes and the system
doesn't reregister and incoming calls are disabled.
Right now the only thing I'm able to do
Le lundi 2 Mai 2005 16:57, Ron Wellsted a écrit :
Guy Decarpentrie wrote:
Hi all,
I try to dial via a Softswitch a number :
exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf)
And my sip.conf
[SIP-OUT]
type=peer
host=10.XX.XX.XX
defaultip=10.XX.XX.XX
disallow=all
Welcome to DSL, the telco didn't do any more tests then required to get
sync for 30 seconds.
Cancel the DSL and get another line. That's about the extent of it, or
at least in Ontario it is, I've had this problem with 5 or 6
connections.
Chad
-Original Message-
From: [EMAIL PROTECTED]
Hello,
Is there a particular order in which codec should be entered in the
oh323.conf file?
I believe that they are put in order of priority. But depending on which
codec is put before another, even if the caller does not support all of
them.
Let me clarify. I have a cisco ATA. When I have this
So I went out and got this IP 500 phone, and see that it has
something called SIP Text messaging. I
can find NO DOCUMENTS out there in Internetland
referring to how this works, or any utility to send it messages. Id love to be able to send
reminders and such to a phone, or group of phones.
On Mon, 2005-05-02 at 09:02 -0500, Matthew Boehm wrote:
Adam Goryachev wrote:
The polycom ip600 and cisco 7960 both have 6 lines available.
Yes, but each of those 6 lines on the 7960 must have their own seperate
SIP username/password. And if you are a small office with 6 7960s, thats 36
On Mon, 2005-05-02 at 08:40 -0600, Michael Welter wrote:
A suggestion was to alter the Called ID Name to the DID number. This
would work for the attendant, but the tenant would like to see the
original Caller ID Name.
Is there an original caller id name ??
You *might* be able to setup some
Title: Choppy Sound on PSTN End
Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux.
Our call routing is like this:
SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice
Kumara Jayaweera wrote:
Hi all,
I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I
don't need incoming calls (no need DIDs). Could someone tell me who is the
best IAX service provider for me? I want unlimited monthly basis or yearly
basis service. my DSL is 128kbps.
Thank
Seems as though the dialpad in SJPhone cannot me used to signal *.
*2 doesn't do anything except play a DTMF in your ear. If you use your
keyboard to send shift-8, 2, all works as expected. Bug report submitted
already.
Cheers
Tim
___
The Box Itself doesn't get a new IP address, the router does. What I'm
looking to do is have the IAX connection re-register every hour or so.
Is this the right idea?
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
Voicepulse is great...
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Monday,
Kumara Jayaweera wrote:
Hi all,
I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I
don't need incoming calls (no need DIDs). Could someone tell me who is the
best IAX service provider for me? I want unlimited monthly basis or yearly
basis service. my DSL is 128kbps.
Kumara Jayaweera wrote:
Hi all,
I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I
don't need incoming calls (no need DIDs). Could someone tell me who is the
best IAX service provider for me? I want unlimited monthly basis or yearly
basis service. my DSL is 128kbps.
On May 1, 2005, at 11:39 AM, Gene Naden wrote:
When we call out from our Asterisk system we consistenly lose the
first
roughly 1500 milliseconds of the audio from the destination. This is
easiest
to demonstrate with a recorded announcement. In other words, Hello
for
example is missing.
Good Day list,
It appears that the CDR is inaccurate, (or I am inaccurate when
reading it) when an attended transfer is conducted with a phones
transfer button
Example
+-+++---
What type of Nortel system? Is it an option or a norstar?
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, May 02, 2005 10:16 AM
Subject: Re: [Asterisk-Users] Asterisk as VM for Nortel System
On May 2, 2005 11:07 am, Matt
I can't seem to be able to make outgoing calls with X100P card. I can
receive calls fine and it picks up the line and sends the tones, but the
telco doesn't recognize them. While the tones are sent I continue to
hear the dial tone on the line when I pick up a parallel. I also cannot
dial from
I have the same problem on a Dell 1850 with a TE410P and have been
attempting to narrow it down. Interrupts don't seem to be a problem and I
have two PRIs from two different suppliers and both have the same
static/chop on the line so it's not the PRI.
The leading suspect at the moment is the RAID
With 128 kbps, you won't make more than 2-3 calls simultaneously...
Ignore this, I read 64kbps. I have corrected this in a follow-up message...
/ducks
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-Original Message-
From: Matt Schulte [mailto:[EMAIL PROTECTED]
Has anyone ever been able to fix this NMI power issue that
the Dell's
have with the TDM cards? Basically locks the machine up when trying to
bring up the module.
I get an NMI the first time I load the module, but the
In all probabilty you will be able to make just one simulatneous call
with that bandwidth, where you need two channels of 64 Kbps each in the
two directions, using Ulaw ( assuming both users are blabbering at the
same time).
You don't need any IAX service providers. You just need a $10 Account
On Mon, May 02, 2005 at 01:02:34PM -0400, Mehmet Tolga Avcioglu wrote:
I can't seem to be able to make outgoing calls with X100P card. I can
receive calls fine and it picks up the line and sends the tones, but the
telco doesn't recognize them. While the tones are sent I continue to
hear
Title: Choppy Sound on PSTN End
I have the exact setup you describe,
SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You
might turn on qualify just to see if the * is seeing network flaws. Keep in
mind, if your using windows, anytime the user starts clicking around, you
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