I have replaced my SPA 3000 with a TDM-400P (which strangely isn't
considered to be a timing source)., and have been keeping the SPA3000 as
a power-down failover. (Home system).
Does anyone know of a device that could be used to replace it in this
purpose?
TIA.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
-- Zap/1-1 is proceeding passing it to
SIP/4053-083189e8
1 Protocol Discriminator: Q.931 (8) len=5
1 Call Ref: len= 2 (reference 13/0xD)
(Terminator)
1 Message type: ALERTING (1)
This is normal, right?
I think so because if I
On Sun, Sep 16, 2007 at 02:38:40AM -0700, Vieri wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
-- Zap/1-1 is proceeding passing it to
SIP/4053-083189e8
1 Protocol Discriminator: Q.931 (8) len=5
1 Call Ref: len= 2 (reference 13/0xD)
(Terminator)
1 Message type:
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Anthony Francis wrote:
Eric ManxPower Wieling wrote:
SIP response 486 is Busy Here according to RFC
3326.
According to http://www.faqs.org/rfcs/rfc3261.html the
SIP client can either send back a 480 or a 486 in case
DND is on:
21.4.18
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
You can probably get an answer to that if you enable
and log debug
messages of Asterisk .
Thanks, I thought that a pri debug was enough but now
I have the missing information:
Sep 16 12:37:28 VERBOSE[19175] logger.c: --
Zap/1-1 is ringing
Sep
On Sun, Sep 16, 2007 at 03:45:15AM -0700, Vieri wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
You can probably get an answer to that if you enable
and log debug
messages of Asterisk .
Thanks, I thought that a pri debug was enough but now
I have the missing information:
Sep 16
On Sun, Sep 16, 2007 at 03:45:15AM -0700, Vieri wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
You can probably get an answer to that if you enable
and log debug
messages of Asterisk .
Thanks, I thought that a pri debug was enough but now
I have the missing information:
Sep 16
Moises Silva wrote:
Open a bug in http://bugs.digium.com/ including all the information
you provided here.
OK, bug id 0010734 created:
http://bugs.digium.com/view.php?id=10734
Also remember to read the bugs guidelines before openning the bug,
this might be already reported.
Regards
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Set busydetect=no
I'll try that tomorrow, thanks.
This really could be the culprit.
In fact, busycount by default is 4 and coincidentally
the Alcatel extension rings 4 times until the
connection is dropped.
So I'm supposing Alcatel is sending back
thx very much Nasir Philipp, I'm gonna try this tomorrow when I'm back at
the server...
however, I wonder if this behavior has changed recently, as I swear [ ;) ]
that this script has been working before...
regards,
michael
On 9/15/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Michael
I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm
using a call file to connect a meetme conference to an AGI script which
plays files using the stream_file method. I have four files which should
play in sequence, though only the first two files actually play. I get
these errors
On Fri, 14 Sep 2007 18:53:53 +, John Meksavan wrote:
Alejandro,
Thanks for replying. I did come by this website before. I was just
wandering, if anybody actually tried Skype with Asterisk. My
experimentation with the Sip Protocol and Asterisk is at end because I
could never get QOS
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Set callprogress=no and busydetect=no in
/etc/asterisk/zapata.conf.
Don't use them. They cause random disconnects.
Thanks, will try.
Vieri wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
You can probably get an answer to that if you enable
and log debug
messages of Asterisk .
Thanks, I thought that a pri debug was enough but now
I have the missing information:
Sep 16 12:37:28 VERBOSE[19175] logger.c: --
Dear Benjamin;
OK friend, things are clear. But now I came to the
same original issue that you asked about it, which is
the ability to stop the log/debug messages into
/var/log/messages.
Same like your situation, the messages is comment (;)
and even the logges are written to the
Did you look at logger.conf?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of bilal ghayyad [EMAIL
PROTECTED]
Sent: Sunday, September 16, 2007 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] stop log/debug messages into
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Kevin P. Fleming wrote:
Paul Hales wrote:
I stand corrected - when I keyed AddQueueMember onto our in-house
production server (1.2.23) I did not see that option. But on my test
environment on my laptop (1.4.10) it's there looking back at me.
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Wai Wu wrote:
Does anyone have this experience? My TCP connection the Asterisk Manager
Interface is chopped off after 15 minutes of operation.
Maybe it is being disconnected because it is responding slowly?
See manager.conf:
; If the device
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Gustavo Gonzalez wrote:
Hi all! I have an issue with TDM400P FXO card. When a call enter into
my IVR and select the proper option, the person that ansswer the call
say your thanks for contact us ... but the caller cant hear this
words because
Michael is bang on -- I've minimized VoIP quality issues at several sites by
reserving sufficient outbound bandwidth. This is especially important if
it's a small office using ADSL.
On 2007-09-16 11:05, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 14 Sep 2007 18:53:53 +, John Meksavan
Hi,
It's probably an echo problem. Try to use fxotune (with new patch) for
determine echo levels.
Try change lines in FXO. For example, use an FXS in some loopback mode.
You can also try to set txgain= - 12 (minus 12) ... this will reduce the
dialtone and DISA will work.
Of course -12 dB will
Is there a way to specify multiple email addresses in voicemail.conf for
a specific user?
I seem to remember that it was possible, but can't remember the
character to separate the email addresses. (I tried '', but that didn't
work...)
later,
PaulH
Michiel van Baak wrote:
On 08:00, Fri 14 Sep 07, H?kan K?llberg wrote:
On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote:
I'm looking for a SIP DECT (cordless) phone for North American
installations. I've heard only of the Siemens Gigaset S450/C450 phones.
Apparently these aren't
I've been trying to post a specific message for the last four or five
days. It's on a specific topic, and I suspect the topic is the reason it
is not being published to the list. Which would suggest that some kind
of keyword filtering is being done, though I've rephrased the message
several
Paul Hales wrote:
Is there a way to specify multiple email addresses in voicemail.conf for
a specific user?
why don't you use the /etc/aliases file for this purpose?
regards
adam
___
Sign up now for AstriCon 2007! September
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Hash: SHA1
Stephen Bosch wrote:
I've been trying to post a specific message for the last four or five
days. It's on a specific topic, and I suspect the topic is the reason it
is not being published to the list. Which would suggest that some kind
of keyword
Matt Riddell wrote:
Stephen Bosch wrote:
I've been trying to post a specific message for the last four or five
days. It's on a specific topic, and I suspect the topic is the reason it
is not being published to the list. Which would suggest that some kind
of keyword filtering is being done,
On Mon, 2007-09-17 at 07:25 +0200, Adam KOSA wrote:
Paul Hales wrote:
Is there a way to specify multiple email addresses in voicemail.conf for
a specific user?
why don't you use the /etc/aliases file for this purpose?
This is more of a question - I don't really need to
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