I'm getting the following error trying to compile zaptel 1.4.0-beta2 on
my RH9 (2.4.20-8smp) system. Can anyone shed some light on this?
zttranscode.c: In function `zt_tc_mmap':
zttranscode.c:387: warning: passing arg 1 of
`remap_page_range_Rsmp_d8cd9cb2' makes pointer from integer with
Hello:
Does anyone know of
a list of milliwatt test numbers for debugging echo?
Specifically I am
looking for a milliwatt test number in Canada, preferrably in a 416 or 905
NPA exchangedifferent carriers would also be niceie. Bell Canada, GT,
Sprint (Now Rogers Telecom)
I calle
e. I found the following that suited my purpose:
Toronto - 416-494-1185
Toronto - 416-439-1185
Oshawa - 905-404-1185
Regards,
Bill
From: William M. Sandiford Sent:
Wednesday, April 05, 2006 1:04 AMTo:
asterisk-users@lists.digium.comSubject: Milliwatt Test Number
List
Hello:
Does anyone
In addition to my earlier
message about an RTP warning, I'm also getting this one a lot. My system
is running just fine, I just keep getting the following
warning message.
Dec 30
10:52:07 WARNING[16732]: app_addon_sql_mysql.c:316 aMYSQL_fetch:
ast_MYSQL_fetch: numFields=7
I really do
I tend to be one of
those kind of guys that likes to eliminate all warnings. Although my
system is running just fine, I keep getting the following
message
Dec 30 10:39:51
WARNING[29172]: rtp.c:779 ast_rtp_make_compatible: Channel 'IAX2/11903-16385'
has no RTP, not doing anything
This m
I just upgraded my
system to the latest svn-trunk
I previously made
extensive use of the SetAccount() function, but now I'm getting the following
error
Dec 29 20:54:08
WARNING[4925]: pbx.c:1679 pbx_extension_helper: No application 'SetAccount' for
extension (voipsubscriber-in, x
Has anyone been
successful getting Auto-Answer by Call-Info to work with the GXP
2000
I have followed the
suggestions in
http://www.voip-info.org/wiki/view/GXP-2000
Specifically I
have:
1. Upgraded to
1.0.1.13, which supposedly supports this feature
2. Set Allow
Auto-Answer by Ca
>On 10/14/2005, "William M. Sandiford" <[EMAIL PROTECTED]>
>wrote:
>
>> I recently upgraded my Asterisk system to the latest CVS-HEAD
>>
>> Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
>> 2005-10-12 13:34:09 UTC
>>
I recently upgraded
my Asterisk system to the latest CVS-HEAD
Asterisk CVS HEAD
built by [EMAIL PROTECTED] on a i686
running Linux on 2005-10-12 13:34:09 UTC
Ever since this
upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is
now proceeding to the next priori
I have tried doing this but I couldn't seem to get the echo canceller to
dynamically turn off properly when it heard the CET while the modems were
initializing. So, what would happen was, I had to turn off the echo
cancellation completely, which allowed the modem calls to work, but the voice
w
Title: Message
Hello:
I've read through
the list archives and found tonnes of threads on this topic but there has been
no definitive answer, so hopefully someone can give me one.
Can a proper 2-way
audio call be established when the UA is behind a NAT firewall and REINVITE is
enabled?
This is not allowed!
>From secion B.1.1 of the Linksys Agreement
Linksys hereby Authorizes and grants any Linksys Service Provider . To
purchase and/or lincense Linksys products from an authorized sourceto
resell and/or redistribute or deploy such Linksys products and services sole
in locating
the correct entry, there is a statement indicating sip show peers will
not reveal sip users.
g
William M. Sandiford wrote:
>Hi Bioz:
>
>I'm having the same problem with Realtime and CVS-HEAD from 4/4/2005.
>
>I haven't found a solution yet, but I have also p
t: Tuesday, April 05, 2005 4:40 PM
To: William M. Sandiford
Subject: RE: [Asterisk-Users] Weird Errors with Realtime and MySQL
> My debug log is empty, and to be quite honest, I have searched for
> how to enable debuggin and can't find it anywhere.
> /var/log/asterisk/debug d
Title: Message
Matthew, thanks for the response.
I connected this error with Realtime based on the fact that I don't have the
problem when I put the config for the UA in the sip.conf file.
My debug log is empty, and to be quite honest, I have searched for how to
enable debuggin and can't find
Hi Bioz:
I'm having the same problem with Realtime and CVS-HEAD from 4/4/2005.
I haven't found a solution yet, but I have also posted this to the list in the
last 24 hours and hopefully someone will help out soon. There was a similar
thread from about a month ago that seemed to solve the probl
Title: Message
For those of you out
there that are Canadian or otherwise interested. The CRTC (Canadian
equivalent of FCC) has released its ruling on 911/E911 for VoIP providers.
In a nutshell it requires all service providers that offer a fixed native
exchange service to provide E911 with
Title: Message
I'm having some
really weird problems with Realtime and MySQL
Some of the
behaviours I have are very similar to a topic that was on the list approximately
a month ago titled "Realtime does not work yet". I have read that full
thread but still can't seem to pinpoint my probl
Title: Message
Hello
All:
Is there a variable
that gives us the IP Address of a SIP caller?
I want to code some
authentication based on IP address into the dialplan.
Thanks,
Bill
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database:
works fine here. I haven't
had an 's' in my CDRs for over several months now.
(continue this on the -users list.)
-Matthew
> From: "William M. Sandiford" <[EMAIL PROTECTED]>
> Reply-To: "Asterisk Developers Mailing List
> "
> Date: Sat, 12
know this isn't the best way but I stopped using 's' and I use _X. on
everything now. It is really stupid to say "That person dialed the number 's'"
or "That phone call's final destination was 's'". That doesn't help anything;
deb
l destination was 's'". That doesn't help anything;
debugging nor billing.
My $0.02
Matthew
William M. Sandiford wrote:
> Hello:
>
> I found a reference to the application SetVarCDR in the following post
> but I don't seem to have this available to me i
Title: Message
Hello:
I found a reference
to the application SetVarCDR in the following post but I don't seem to have this
available to me in my version of *.
http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html
My version of * is
CVS-HEAD-03/10/05-18:42:35
I would
k out of
thequeueand go to voicemail
On Sat, 6 Nov 2004, William M. Sandiford wrote:
> Excuse the newbie nature of the question, but can you elaborate a
> little further. Sorry...I am pretty new
There is a block in the queues.conf.sample file in the Asterisk
distribution that reads:
: [Asterisk-Users] Giving users the ability to break out of
thequeue and go to voicemail
On Sat, 6 Nov 2004, William M. Sandiford wrote:
> Hello All:
>
> I need some help. I am trying to configure * so that users that are
> placed in a call are able to break out of the que
his?
Bill
-Original Message-
From: Matt Gibson [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 06, 2004 2:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Giving users the ability to break out of
thequeue and go to voicemail
William M.
Hello All:
I need some help. I am trying to configure * so that users that are
placed in a call are able to break out of the queue and go to voicemail
if they no longer wish to wait in the queue. I read the cmd options for
the Queue command and I had figured that either option h or H would do
th
Title: Message
Hello
All:
I am trying to setup
an queue for my technical support department and I am having a really weird
problem. For some reason my queue only accepts one call. When the
second call comes in I get a warning message and the user is
disconnected.
The messages when
t
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