Displaying 20 results from an estimated 5000 matches similar to: "DTMF issues..."
2007 Oct 15
1
Answering Machine Detection
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
Here is the output of a call on my office server:
-- Attempting call on Local/0445540881644 at CC2 for
2012 Aug 02
1
Originate call from cli does not work for SIP line...
I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to
dial from the CLI using the originate command or use an AMI connection
to originate a call I get the following error:
originate SIP/protel-out/0445540881644 application playback tt-monkeys
WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received
2011 Feb 02
0
SIP Originate on 1.8.X
I am having a problem trying to use originate from the CLI on Asterisk
1.8.2.3. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:
pbxoficina*CLI> originate SIP/protel-out/0445540881644 application playback tt-monkeys
[Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048
handle_response_invite: Received response:
2008 Dec 05
2
async agi question
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
2008 Feb 01
2
Asterisk 1.4.17 and Teliax DTMF
I am having a problem with DTMF when sending calls through Teliax
(SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working. The problem always happens when a user is
trying to call a conference system. They simply cannot get into the
conference because DTMF is not understood. If I dial from a land line I
can get in with no problems.
Any tweaks recommended
2009 Apr 08
3
Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Hi,
I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL))
with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2
version. The outcoming calls are ok, but with incoming call i have an error:
ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi
Frequency Cycle Timeout, R2 State =
Seize ACK Transmitted, MF state = Category Request Transmitted,
2009 Mar 30
1
The Redirect hangups the call while playing a file
Hi,
I'm bringing this discussion here from
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
about how to manage stopping a playback on a extension previously launched
with AsyncAGI and redirecting the call to another exension.
If I make the Redirect without a playback, the Redirect works:
http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd
But if I make the
2007 Nov 05
1
PRI dialout problem with some numbers...
I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico.
This is really the first server I have used with PRI in Mexico as we
normally use MFC/R2. Everything seems to be working except that some
numbers always seem to be busy when you dial them. All these numbers
belong to different phone companies. I know that with R2 this problem
is present if you have a "#define
2003 Sep 25
1
Per-directory "create mask"?
Hi all
Please CC me any replies.
I am currently running samba 2.2.8a under Slackware 8.1 (kernel 2.4.18) to
provide fileserver services to a number of windows 2000 boxes via a single
"resources" share. One of the things stored in this share is a collection
of Protel libraries used in electronic circuit design.
The problem is that protel implements its own access control mechanism for
2007 Dec 02
2
Answering Machine Detection
If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting?
---- dave cantera
2006 Jul 28
0
mistake drawing the bottom of window in PROTEL (regression)
Hi everyone:
I use the protel CAD software. In early versions this work well, but in this
version i got a little problem
Whe execute in terminal appear this into the screen, no other message no
fixme, no warning only this
fixme:int:WIN87_WinEm87Info (0x7e285aee,12), stub !
But the things that must appear at the botom of window have a extremly big
bottom border. so big that i cant see the
2008 Apr 17
2
G729 license count...
I need a refresher course on how many licenses I need to buy. I have
an Asterisk server that receives calls by SIP (G729) and then sends them
to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if
the license is per channel or per call so I do not know if I need 32 or
64 licenses for this application. Could anyone please remind me?
--
Telecomunicaciones Abiertas de M?xico
2009 May 26
1
STUN setting in Asterisk 1.6.X
I have been trying out several stun servers with Asterisk 1.6.0.9 and
1.6.1.0 and I keep getting the following message:
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]:
2008 Jun 11
2
Zaptel timer failure
Hi all,
on a new installation, HP Xeon at 1,83 with 2.6.18-amd64 image from Debian
Etch with Asterisk 1.4.20.1 and zaptel 1.4.11 -both tar.gz from digium-
we get:
[Jun 11 14:54:29] ERROR[6686]: asterisk.c:2952 main: You have Zaptel
built and drivers loaded, but the Zaptel timer test failed to set
ZT_TIMERCONFIG to 160.
There is no telephony card in the server, we want to use ztdummy for
2008 Feb 06
3
R2 with Alestra in Mexico...
I am trying to set up Astunicall 1.4.16 with a link from Alestra in
Mexico City. I have done everything I usually do for other links in
Mexico but this one simply will not send or receive calls. I just get
Protocol error.
Anyone has any experience with R2 and Alestra?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
2004 May 07
0
Authentication failure
Hi,
I'm configuring samba-3.0.2a-Debian. I have it configured with LDAP.
I get samba decline access for a W98 clinet when using user authentication
(NT domain), and the following entries get into /var/log/smb/machine.log:
--------------------------------------------------------------------
[2004/05/07 13:46:07, 3] smbd/sesssetup.c:reply_sesssetup_and_X(772)
Domain=[WORKGROUP]
2006 Apr 11
4
Unable to run application from share
I'm not sure if this is a samba problem or just a problem with the app.
I'm unable to an application from a shared drive "\\ucd01\protel\". I've mapped the network drive "k:\protel\"and the apps ini file points to it but gives me an error message suggesting it can't see the drive. XP can however and I can browse it in explorer.
A strange thing is that when I
2006 Apr 06
1
Fwd: RE: Not able to join domain
Sorry about the direct post....
>Date: Thu, 06 Apr 2006 08:38:39 -0500
>To: "Chris Boyd" <Chris.Boyd@usit.ie>
>From: Eric Hines <eehines@comcast.net>
>Subject: RE: [Samba] Not able to join domain
>
>At 04/06/06 08:13, you wrote:
>>I've tried that and now I get "Access denied" instead "cannot find
>>user". Also I'm
2011 Aug 23
3
[LLVMdev] git Status
greened at obbligato.org (David A. Greene) writes:
> Actually git pull can sometimes get you into trouble. Probably git
> fetch / git rebase is the better combination for LLVM.
I don't get it.
Doesn't "git pull --rebase" do exactly a fetch followed by a rebase?
--
Matthieu Moy
http://www-verimag.imag.fr/~moy/
2006 Oct 26
6
Client-identifier option in PXE search order
Hello,
Is there any plan to use the client identifier field as part of the PXE
search order ?
This will be helpful when using PXELINUX on Infiniband network which do
not have 6 byte Ethernet MAC address.
Thanks,
_________________________________________________
Nir Gal | +972-9-9717685 (o) | +972-54-499-6633 (m)
Manager, Customer Support
Voltaire - The Grid Backbone
www.voltaire.com