Displaying 20 results from an estimated 3000 matches similar to: "MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped"
2004 Sep 20
5
iax2_read: I should never be called
Skipped content of type multipart/mixed-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 252 bytes
Desc: OpenPGP digital signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040920/0629df7b/signature-0001.pgp
2010 Aug 06
1
fix for unsafe ssl options
Hi Niv,
Thanks for your comments. I'm CC'ing the patch author.
On Wed, Aug 04, 2010 at 05:37:44PM +0200, Niv Sardi wrote:
> Do we really need that many ?
> http://www.google.com/codesearch/p?hl=en#5KTrgOW2hXs/pub/nslu2/sources/vsftpd-2.0.4.tar.gz%7CXknrlk4c3C4/vsftpd-2.0.4/ssl.c&q=SSL_CTX_set_cipher_list
>
> vsftpd seems to only be including "DES-CBC3-SHA"
>
2004 Sep 14
2
Spawn extension.....exited non-zero
I am recieving inbound calls to my asterisk box over IAX.
And getting "spawn extension....exited non-zero" errors.
Im not entirely sure what this means, and would appreciate any help in
fixing my problem.
FYI:
********** is the inbound phone number
x.x.x.x is a remote asterisk box calling my own asterisk box.
When I choose it to dial an internal extension using this dialplan:
exten
2010 Aug 04
2
fix for unsafe ssl options
While he was at it justdave also fixed the ssl options.
https://trac.xiph.org/ticket/1718
2007 Jun 08
0
No/unknown event '0' on timer
Hey guys,
I'm currently running Asterisk 1.2.18 Under Mandriva Linux. Three
Facilities are hooked together via IAX2 (Trunked) over a OpenVPN
connection on a 10mbit (uplink/downlink) internet connection. I was
parked for around thirty seconds at a remote facility. All of a sudden,
the call drops. The log entry was:
Jun 8 11:34:14 NOTICE[10458]: channel.c:1918 ast_read: No/unknown
2007 Mar 28
2
Meetme cant handle more than 5 users in a call?? hmmmm
Meetme cant handle more than 5 users in a call?? Hmmmm
http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/
hmmm I'm all for commercializing a product, but this FUD from Fonality
seems to be taking it just a little too far
Regards,
Dean Collins
Cognation Pty Ltd
dean@cognation.net
<mailto:dean@cognation.net> +1-212-203-4357 Ph
-------------- next
2007 May 19
1
Call someone to instantly join conference using MeetMe
Hi,
I was just wondering how would the application be where the Asterisk calls a
number and that number joins the conference as soon as the call connects.
There would be only one conference already defined in meetme.conf and there
is one person already joined the conference. Currently MeetMe requires a
person dialing into it and the joining the conference. How could this be
done using MeetMe or
2006 Nov 23
0
festival problem using IAX (chan_iax2.c:2995 iax2_read)
Hi All,
I'm having a problem after reinstalling the operating system.
Festival works fine for SIP, but when IAX users are calling the same
extension they don't hear the festival and I see the next message on
console:
NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called!
I googled and couldn't find a solution, if somebody can help....
neobase*CLI>
2005 Oct 12
5
delays with IAX2 and Meetme
Hi there
I am using IAX2 softphones dialing into meetme conferences. I also have
jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
having is that as soon as there is a delay from a participant, then the
delay continues until the participant hangs up and dials in again. When
dialing in again the delay seems to go.
It seems to me as though as soon as the server registers
2007 Apr 26
3
Two devices registrating same extension
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: Esta =?ISO-8859-1?Q?=E9?= uma parte de mensagem
assinada digitalmente
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/aab9f324/attachment.pgp
2009 Nov 18
0
Asterisk 1.2.18 and meetme causing Audio bleeds
Lookin for anyone who has experienced an issue similar to this. It's quite
baffling as I'm unable to locate much help when it comes to debugging such
an audio oddity.
I'm currently running Asterisk 1.2.18 with a T1/E1 PRI.
To cause the audio bleed (is audio bleed actually what I should even call
this?): I create a meetme conference and have a few people call in to it.
Once
2007 Nov 06
4
MeetMe CPU resources
Hello,
We would like to have a conference with 15 users aprox. We think that
Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running.
We wonder if somebody has some other experience, good or bad.
We will use Asterisk 1.2 (it is a small and short project for only
this).
Thanks!
--
Carles Pina i Estany GPG id: 0x8CBDAE64
http://pinux.info Manresa - Barcelona
2004 Aug 31
0
MP3Player strange error
Hi all!
I downloaded right mpg123, chabged path to mpg123 binary in
app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But
MP3Player refuses to do properly:
-- Accepting AUTHENTICATED call from x.x.x.x, requested format =
1024, actual format = 1024
-- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack
-- Executing
2005 Aug 18
0
MP3Player cmd issue
I am running CVS HEAD (on a Linux-PPC machine.)
My current dialplan generates an error at the console in asterisk
when I attempt to issue the MP3Player command -- I can't figure out
why it's not playing the actual audio file?
The rest of the dialplan works great.
Here's what I see in the console:
-- Executing MP3Player("IAX2/income-in-01@IP",
2004 May 18
3
call announce? in MeetMe?
has anyone done caller announce in MeetMe's before?
Dave P
>>> brian@bkw.org 5/18/2004 5:50:49 PM >>>
With multiple parking lots you can give each person their own lot thus
exten
800 for everyone will connect them with just their call passing the lot
name
which you know for X customer.
bkw
----- Original Message -----
From: "Andrew Kohlsmith"
2004 Jul 23
0
qudBRI and transfering calls with the latest RC2.
I'm trying the latest bri 0.1.0 RC2 drivers.
In announce I see implementation of so long waited Transfer feature.
But I can't make it work.
When the person who is making transfer after talking with second party press
"R" second time to establish 3 way call
the person to which call supposed to be transfered being disconnected.
Any ideas whats wrong?
Thanks,
Dmitry
2003 Apr 25
1
MeetMe over IAX2 Test
We want to test capacity of our MeetMe room. The thing that is distinct
about this is that the incoming line is being delivered IAX2 to our server
across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test
both the VoIP Gateway and the MeetMe room performance.
You can reach our MeetMe room directly at 1-301-561-9229
If you want to test with us we're thinking maybe 9pm
2004 Nov 23
1
IAX2->SIP->meetme = ZOMBIE
Hi all,
I'm experiencing a problem with SIP channels going ZOmBIE after the
following sequence of events:
- IAX2 client calls SIP client
- SIP client consultive transfers (using sip REFER) the call to a MeetMe
extension, and hangs up.
At this point, the IAX2 client will indeed be in the meetme room, but a
'show channels' at the * CLI reveals that the SIP channels that were
involved
2011 Apr 19
3
No voice in MeetMe for SIP with AGI_BACKGROUND
Hello List,
I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.
2004 Aug 16
1
Is "Meetme" a generic term?
Just a trivial question: was the term "Meetme" invented for Asterisk
as something like a brand name for its conferencing? Or was it an
existing generic term for dial-in conferencing?
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org