Steven Critchfield wrote:
On Sat, 2003-10-04 at 12:02, Lists wrote:
But would you then not be able to use caller id from the telco?
CallerID on an analog line is sent between the first and second ring
normally. So if the requester wants callerid and first ring answer, he
will have to move to
Linus Surguy wrote:
A little off-topic, but does anyone know if it is possible to disable call
waiting on the Grandstream phones? I can't find anything in the
configuration that would allow this.
Linus
No there is no way to disable it..
I tried using incominglimt=1 and outgoinglimit=1 but
Fats Neutron wrote:
I have tried loads of configurations but I cannot get it to work.
I basically have three computers and want to use soft phones on two of them
to connect to asterisk so I can use them instead of dialling into my X100P
card as the phone bill is getting bigger.
I assume that I
Is that all systems? The IKP, 2000, and 2400?
- Original Message -
From: Doug Dimick [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, September 21, 2003 11:32 AM
Subject: [Asterisk-Users] Re: how many production systems are there?
On Saturday 20 September 2003 14:29, Steve
I had a similar problem when using an AS5300 with SIP. I'm not familiar
with the 7940, but on the AS5300 there is a Voice-Activity-Detection
setting. Disabling the VAD on the AS5300 corrected the SIP music on hold
problem.
Russ Beaupre, P.E.
- Original Message -
From: Babak Pasdar
Sorry about this off-topic question... I want to
know if the second ethernet port on the Grandstream 102 phone works as a bridge
to connect from there to a PC. Do
I need two ethernet jacks to connect a phone and a
PC, or this phone let me connect both with only one? Thanks in
advance!
On Sat, 4 Oct 2003, James Sharp wrote:
Actually, if this was to be done, it might be an idea to do it with DNS, so
client machines would just do
Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS
system would resolve which machine is the correct target - no cleverness at
all required
On Sat, 4 Oct 2003, Steven Critchfield wrote:
On Sat, 2003-10-04 at 12:02, Lists wrote:
But would you then not be able to use caller id from the telco?
CallerID on an analog line is sent between the first and second ring
normally. So if the requester wants callerid and first ring answer, he
Nicolas Gudino wrote:
Sorry about this off-topic question... I want to know if the second
ethernet port on the Grandstream 102 phone works as a bridge to
connect from there to a PC. Do
I need two ethernet jacks to connect a phone and a PC, or this phone
let me connect both with only one?
On Sat, 4 Oct 2003, James Sharp wrote:
Actually, if this was to be done, it might be an idea to do it with
DNS, so
client machines would just do
Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the
DNS
system would resolve which machine is the correct target - no
cleverness at
all
I am trying a scenerio where the * will take the email and
mailbox number from the Mysql database for sendming mail to a voicemail user. I
have seen vmdb.sql file but is not able to determine its use.
You can't anymore MySQL was ripped from Asterisk because the client
libs are GPL.
I
When I place a call using Iconnecthere
or Nikotel as my sip provider, I hear no call
progress such as ringback when making a call. If I program the SIP phone to directly
access iconnect or nikotel
I do hear ringing when the outbound call is placed. Does anyone else have this problem or
Thanks for the response Russ, I had to disable VAD to get rid of the
annoying message that shows up on the console. So on both phones VAD is
disabled. Subsequent to your response I also disabled in phone.conf:
silencesupression=yes
Same result:
No ringing / No Music On Hold
No Music on
Typically you run a cable into the phone, then a cable out of the phone
into the computer, it appears to just be a bridge. It also works fine to
run the cable into the phone and then a cable into another phone and then
into a computer.
Michael
On Sun, 5 Oct 2003, Nicolas Gudino wrote:
Thanks Paul!
You are correct... even though it says it will not change your
IAXTEL password, it did...
G
At 07:23 PM 10/4/2003, you wrote:
I had exactly the same problem. You change your password at the gnuphone
website and it says it doesn't update your IAXTEL password, but it does! I
changed
Mark Spencer wrote:
It won't cause any sort of serious problem, but you are getting it with an
unusually high frequency. What's particularly interesting is that it
occurs almost exactly once per minute, on the minute. This would seem to
suggest that you have some hardware in your system which,
a.) In an effort to push Linux's poor job of handling interrupt
latancy, which makes it a not so good choice for real time systems
that the TE410 driver is simply pushing the limits a little.
Which is all right by me :)
Well, Zaptel needs to service interrupts once every 1000 microseconds.
Ken Godee wrote:
I'm trying to figure out how to start *.
Rh7.3,CVS,TE410P,TA750
If I just try the way the docs spell it out
/usr/sbin/asterisk -vvvc it fails..
Gezzz, must just be me
but I'll get an email in the list incase anyone else can use in the
future. Right or wrong, it's
It is that type of mechanism that enum uses and yes it was to solve a
similar goal, but in this case you need a 'route server' type system - in
particular as this is for IP routing of PSTN end points not on an IP
network.
A discussion about this came up a while ago. I suggested something
Hi all...
How can i register wit nufone i was serching at its pages... and
I never find how to get register...
Thanks.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Like what PEARS (php libs) do for db backends?
Matteo.
Garry Adkins wrote:
I am trying a scenerio where the * will take the email and mailbox
number from the Mysql database for sendming mail to a voicemail user. I
have seen vmdb.sql file but is not able to determine its use.
You can't
Email [EMAIL PROTECTED] to get setup.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alvaro Parres
Sent: Sunday, October 05, 2003 2:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] HOW TO GET REGISTER WITH NUFONE??
Hi all...
How can i register
Talk to Nufone sales, to get an account setup with them.
[EMAIL PROTECTED]
quote who=Alvaro Parres
Hi all...
How can i register wit nufone i was serching at its pages... and
I never find how to get register...
Thanks.
___
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead of
the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
Hi... some one can tell me his comments about VoicePulse Services...
for Pre-Paid long distance..
Alvaro parres
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On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How
Hi..
I'm having the next problem... with the busy detect = yes...
If i have it... The * it hang up the calls when they are
active... ( Incoming ant Outgoings calls).
If i haven't it * Doesn't detect when some one hang up and
never close the channels...
WHAT I CAN DO
When I place a call using Iconnecthere
or Nikotel as my sip provider, I hear no call
progress such as ringback when making a call. If I program the SIP phone to directly
access iconnect or nikotel
I do hear ringing when the outbound call is placed. Does anyone else have this problem or
What about libdbi? It already has the pluggable driver system, so
thats half the work done already.
Its also licensed under the LGPL instead of the GPL, which if I have
understood the discussions about
the GPL issues, should resolve problems of linking to that code.
Support for sqlite is cool
What is the
Exten = .Dial(
Line from your extensions.conf?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
7:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Ringback on Iconnect
When
I have tried both of
these:
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED]||r
-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED]
Sent: Sunday, October 05, 2003
7:56 PM
To:
[EMAIL PROTECTED]
Subject: RE:
Try
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)
or
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],90,r)
It should work
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October 05, 2003
9:04 PM
Title: Message
Look into AGI, and the associated AGI interface perl
module. The interface is super easy to use in perl to gather digits
and perform your own logic.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Sunday, October 05, 2003
5:54 PM
Changed my conf
file to:
exten =
_1XX,1,Dial,SIP/[EMAIL PROTECTED],90,r
still no ringback
-Original Message-
From: Andrew Joakimsen
[mailto:[EMAIL PROTECTED]
Sent: Sunday, October 05, 2003
9:20 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No
Ringback on
Then remove the ,r and see if it works.
Have you setup your indications.conf? Do you get any messages from asterisk
CLI? What device are you using?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kevin
Sent: Sunday, October
05, 2003 11:01
hi,
I have lately acquired a TE410P. The problem i face currently is
that the span gets(or doesnt gets) UP in an uneven fashion i.e i have to
load-unload the modules, wait for sometime and then start the
application i dont know but this execise works well for me (though
not neccassary
Rahul Arvind Jadhav wrote:
hi,
I have lately acquired a TE410P. The problem i face currently is that
the span gets(or doesnt gets) UP in an uneven fashion i.e i have to
load-unload the modules, wait for sometime and then start the
application i dont know but this execise works well for
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