Displaying 20 results from an estimated 30000 matches similar to: "Meetme and IAX"
2004 Sep 24
2
VICIDIAL and IAX
Hello everybody,
I would like to know if there is a support of IAX in vicidial.
I want to make predictive dialing use vicidial using IAX soft phones.
Thanks in advance
Lamine
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2003 Jul 22
0
IAX / MeetMe problem
Greetings,
I have a somewhat unique (I think) configuration that I am testing
involving MeetMe conferencing and have encountered a problem that I'm not
quite sure how to solve. Here is a brief description of my setup for the
background.
I wanted to offer the ability for users to mute and unmute themselves while
in a conference. If they enter a conference as monitor only, they are
2005 May 12
3
How to decrease Asterisk load
Hi everybody,
I would like to decrease the load of my asterisk server. Could someone
recommend me a solution? I have thought about a hardware component that
would do some tasks as compression/decompression or codec translations but
wonder if such a solution exist.
Thanks for any suggestion
Lamine
2004 May 18
1
Dial and MeetMe on the same channel
Hello everybody,
I would like to know whether it is possible to run Dial and MeetMe
commands simultaneoously on the same channel.
I am using a C AGI as below but it seems to me that only the first
command that is called in the agi is executed.
...........
// Pr?paration de la commande pour l'appel du client
fprintf(stderr,"%s%s",numtocall," is the number to
2004 May 19
3
Call recording between SIP phones
Hi everybody,
I have been searching around for days on how to record calls between SIP
phones.Could someone tell me whether it is possible? The Record command
doesn't seem to work during a call.
Thanks
Lamine
2006 Apr 20
1
MeetMe: lots of buffer overruns/underruns when connecting over IAX
Hello,
Situation: I've got two asterisk 1.2.4 servers, connected to each
other over the internet with IAX2 with about 20msec delay.
One of the servers is hosting MeetMe. It's working fine as long as
only SIP phones connected to the meetme server participate in the
conference. As soon as a participant using IAX2 is connecting, lots
and lots of buffer overruns and underruns are
2005 Jul 06
0
Dropped calls if transferred across servers into MeetMe with mobile source
I have an application where calls come into an *box from a DID
provider, and may be transferred to a meetme conference on another
*box (the call is released by the first *box after transfer).
These are ulaw IAX channel calls, and if the source is from a Verizon
or Nextel mobile phone to the DID (other carriers not tested), the
call drops about 2-3 minutes after it joined the meetme
2006 Mar 16
1
MeetMe - Causes * to crash :/
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
"conf-onlyperson". This has been seen with the MeetMe participant connecting
via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't
seen it).
The box is *
2006 Mar 23
1
RE: MeetMe freezes machine with Junghanns
Dollars to donuts it is related to these two posts, but no one seems to know
where or why it happens - this issue doesn't seem to be related to one
specific piece of hardware:
Post 1)
*********************************************************
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while
2006 Jan 31
1
meetme and dtmf
Hi all,
I'm experiencing a problem with meetme i can't resolve.
This is my scenario:
A iax client, say IaxComm, make a call through a zap channel. When it
answers it is tranfered to a conference room.
Then the iax client make a second call though a second zap channel, at
the other side there is an IVR. Iax client send some dtmf to the IVR
then it transfers the IVR to the previos
2006 Mar 18
0
Re: Server freeze with meetme and sip GSM users
Hi Brent
> Anyone ever seen MeetMe cause * to crash? Specifically, it happens
> consistantly if someone begins to enter a conference and then decides to
> hangup while Allison is introducing them - like playing back
> "conf-onlyperson". This has been seen with the MeetMe participant
> connecting via IAX and SIP (not saying it doesn't happen with Zap, just
> that I
2005 Sep 09
0
Transferred calls dropping out of MeetMe
I'm taking inbound calls on an * server, then transferring them to a
second * server where they join a MeetMe conference. If I have
'notransfer=yes' set on the first * server it works fine, but if I
allow the transfer (call then shifts to be between the DID provider
and the second server), the call is dropped 3-5 minutes later.
There is no firewall on my end, and the two
2005 Aug 27
0
how can I reduce delays in meetme with zap channels
My boss is complaining that the delay between speaking and hearing in a
meetme conference is noticeable and doesn't want to roll out our system
until I can eliminate the delay.
Personally, I don't think the delay is significant, but I don't sign his
check.
The system consist of 3 1u's, each with a single quad t1 card. Each card
has 2 t1's running NFAS.
The "t1
2004 Dec 12
0
MeetMe performance
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
conference, the audio is very delayed, choppy and segmented -- totally
unusable.
At the
2010 Jan 22
0
Meetme conferencing - large deployment SIP or ZAP?
I've been asked by my company to setup a conferencing system to support up
to 400 people on a conference calls, where all users will be dialling in
frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two
questions in relation to this:-
For Meetme conferences is it better to have all participants to dial in via
SIP provider terminating to Asterisk via SIP/IAX, or use
2005 Oct 04
0
Three-way calling over SIP and IAX using softphone
Hi guys,
Does anyone know of a way where I can bring a third person in on my
conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM
and am speaking to someone now I want to quickly bring another SIP or IAX
extension into this call so the three of us can speak to each other.
I know I could do this by transfering the first person into a meetme then
calling the second
2005 Jan 01
3
Announcements via IAX phones
Hello--
What I'd like to do:
Use IAX softphones running on computers, in Auto-answer mode, with sound
going to speakers, as a sort of public announcement system.
What isn't working:
Well, my first experiment was to set up the MeetMe system described on
the Wiki...
This works fine for voice announcements. You pick up a phone, dial the
right extension, and an agi is fired up to put files
2005 Jan 07
7
Problem with call pickup
I have configured call pickup, and this works fine.
Although there are 2 problems, perhaps anyone would know a solution to this;
- When I pickup a call from another set, the *8 code keeps being displayed
in my screen (Snom 220).
I would like it to show the phonenumber of the person calling me.
- When a caller that I've answered through Call-Pickup disconnects, my phone
does not close
2005 Feb 02
1
Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy = ringall
context = from-pstn
timeout = 15
retry = 5
maxlen = 0
announce-holdtime = no
announce-frequency = 30
announce-holdtime = yes
monitor-format = gsm|wav|wav49
monitor-join = yes
eventwhencalled = yes
member => Agent/1000
2007 Apr 26
2
MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
We upgraded our asterisk server to 1.2.18 last night to pick up the
security update. Since then, any calls coming in on IAX2 links get
dropped if they try to enter a MeetMe conference room.
The log shows this:
Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should
never be called! Hanging up.
I've temporarily worked around it by switching our inbound provider to
use SIP