can anybody identify why the CLI is issuing a failure message
while debug shows everything is fine
this makes no sense to me.
also, why is the username being updated? this has got to be wrong
from CLI
-- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '52221' at
Got that from grandstream, and testing it for a couple of things
Greg
At 12:47 PM 12/24/04, you wrote:
Greg - Cirelle Enterprises wrote:
-- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221
Greg,
Completely unrelated to your current query. Your logs show that your
BT100 is running
At 06:24 AM 12/25/04, you wrote:
On Sat, 25 Dec 2004, John Bittner wrote:
Anyone having any problems with CVS-HEAD-12/24/04-15:59:15
and ALERT_INFO
I have a system setup with polycom phones configured to auto
answer on internal calls. When we upgraded to the latest CVS
the auto answer stopped
does anybody know what these log messages mean?
what ever it is, asterisk needed a restart to become active
again. (server was not rebooted and remained live to ssh
and other network functions.)
no outgoing calls can be made.
The system was just sitting idle over night and trying to make
a call
At 05:49 PM 12/27/04, you wrote:
I have a similar problem with my *.Works fine but after some number of
hours, nothing works with no apparent reason. Restarting * fixes
everything.
I hope someone comes up with some suggestions!
Norm Z
I just downloaded a new cvs to see if that helps,
At 06:12 PM 12/27/04, you wrote:
James Moran wrote:
I just updated my asterisk box and now it's giving me this error I looked
it up on the internet found no solutions
any other information that you need please ask.
[EMAIL PROTECTED] root]# modprobe wcfxo
At 01:37 PM 12/27/04, you wrote:
Hello *'s,
Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot find
this option.On wiki i found this.
To enable this, you need to edit the Makefile in the channels directory of
your source tree and enable MYSQL_FRIENDS. This enables database
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21
or
10 7 * * * root
At 09:19 AM 12/30/04, you wrote:
Greg - Cirelle Enterprises wrote:
from voip-info wiki
Asterisk automatic daily restart
To automatically restart Asterisk you can add something like this to cron
# Restart Asterisk PBX once a day to prevent any problems from piling up
10 7 * * * root /usr/sbin
At 11:00 AM 12/30/04, you wrote:
I wouldn't say it's unstable... these boxes all run res_perl and reload
100's of times a day. It all depends on if you know what the hell you're
doing.
bkw
why are they reloading 100's of times a day??
greg
___
At 02:17 PM 1/3/05, you wrote:
I m about to purchase an adaptor for a POTS data modem and was looking at
the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do
these work well? Anyone have a suggestion on which model of the Sipura I
should get? Does one work better with *
At 03:25 PM 1/3/05 -0500, you wrote:
Unfortunately that makes Asterisk installs for small businesses more
expensive
than necessary. At US$500 for a T100P and US$300ish for a channel bank (FXS
only, FXO is significantly more expensive!) plus your time and system for an
Asterisk install it raises
At 11:34 AM 1/4/05, you wrote:
On Tue, 4 Jan 2005 10:08:27 -0500, Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial
At 12:29 PM 1/4/05, you wrote:
Greg - Cirelle Enterprises wrote:
When you try to sell the asterisk system, you have to compete with that and
frankly, all the people want is to make phone calls.
Mention voice over ip and eyebrows raise, I've heard of that, but in
reality
nobody cares how
At 12:50 PM 1/4/05, you wrote:
Sipura SPA 3000... forget the channel bank and PRI card. Buy a PRI card
and ebay the SPAs when you arte ready to move from POTS to PRI, or
better yet, forget both and find an ITSP that can offer QoS (private
line!!!) and interface with *
Talkswitch? Get on the VoIP
do a google search for
tdm400p hardware problems (fix)
This is a problem with the tdm card and driver
If you are using the older zaptel software the
file referenced in the doc is wcfxs.c if you
are using the cvs version the wcfxs file needs
to be replaced with wctdm.c also the line number
2127 is
At 02:07 AM 11/28/04, you wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
Regards
Greg
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Does anybody have the settings to allow the BT 100
to properly display the caller id text or replace
the text with the numeric value?
Regards
Greg Cirino
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just testing
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At 06:24 PM 12/4/04, you wrote:
Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
Thanks Doug, will try that
Greg
I seem to be missin the save dialplan command in
asterisk 1.0.2, I have been searching for info
but all I get is how to use it.
Anybody have any info on this?
Regards
Greg Cirino
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analog phone. (Example: some central offices don't like dtmf tones
within xxx milliseconds after going off-hook. You'll get wrong
numbers, etc. Insert the 'w' option in your Dial statement to
delay those dtmf tones a little bit.) To be a little sneaky,
We had one line, it happened to be a
bt100 1.0.5.18 (same results with previous versions)
an outside call comes in via fxo to an extension hosted
by a sipura 2k. the call is parked and is able to be
picked up (un parked) on a second extension hosted on
a completely different xipura 2k. the call is then reparked
and an attempt is made
At 06:29 PM 12/9/04, you wrote:
www.grandstream.com/BETATEST
- Original Message -
From: Mark Willis [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Thursday, December 09, 2004 11:36 PM
Subject: RE: [Asterisk-Users] BT-100 Transfer!!
Does anyone know why the bt100 cannot park and pickup
a parked call?
attendant announces the call is parked at extension 701
but the call cannot be retrieved by any other phone.
also, the bt100 constantly rings when the phone is
hung up after parking.
anyone experienced this?
using the basic
At 08:57 AM 12/10/04, you wrote:
On Friday 10 December 2004 13:23, Greg - Cirelle Enterprises wrote:
Does anyone know why the bt100 cannot park and pickup
a parked call?
attendant announces the call is parked at extension 701
but the call cannot be retrieved by any other phone.
also
At 07:05 AM 12/10/04, you wrote:
greg wrote:
Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel
Diego Aguirre wrote:
I use atendent transfer in Asterisk!!!
ok, the # extension key combo
too bad the 3 buttons cannot be programmed to emulate
the functions to make it work.
At 09:24 AM 12/10/04, you wrote:
No,
i don't use the # key...
100 cal to 200 (BT-100), 200 press flash then 200 call to 300.. 200 talk
to 300 and press transfer key (or hangup), now 100 talk to 300.
the same is useful for Handytone ATA 286...
sorry my english, this is not my language...
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
greg
Regards
Greg Cirino
___
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603-425-2221
www.cirelle.com Web Application Development Design
www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster
At 08:19 AM 12/13/04, you wrote:
On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote:
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
How about wctdm.c ?
--
Dave Cotton [EMAIL PROTECTED]
Not sure what that is supposed to do but it
sure don't
At 09:32 AM 12/13/04, you wrote:
Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never works. The best I've gotten is the MySQL query to
execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
libpri asterisk asterisk-addons
At 04:59 PM 12/13/04, you wrote:
Can anyone give me some recommendations for IP phones that work well with
Asterisk?
I'm hoping for something not much more then $100 bux or so.
grandstream bt100 will work 100
Also does vonage service work directly through Asterisk or would I have to
use their
Not being an asterisk expert, but having been around
the block once or twice when it comes to data and the
like, I have made some observations based on the examples
given on voip-info.org Sip configs.
it appears there is an adjustment to be made in
the sip_buddies example table:
name
Although set
It's been hours since I've seen a post from this list
Must be broken again.
Regards
Greg Cirino
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603-425-2221
www.cirelle.com Web Application Development Design
www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster
www.cedata.com
Some of the others you mentioned, name etc, can be increased. But most of
those options that call for 'Yes', 'No' or NULL can all be 1 char wide.
-Matthew
Thanks Matthew,
greg
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Sure, why not? You know, like how your PHB emails you to let you know the
mail server is down.
--
Tracy Reedhttp://copilotcom.com
PHB
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At 01:53 PM 12/15/04, you wrote:
anynody knows if I Can install and run Asterisk under Free BSD?
/usr/ports/net/asterisk
randy
several months ago, the port for asterisk was not working because
of a security failure in h323. don't know if it has changed or not
or if the security issue in h323 has
At 04:48 AM 12/16/04, you wrote:
Dear Members,
I am searching for a new PBX for the company. My choice is Astrisk. My Boss
wants background music via all the telephones.
If you don't mind my asking, what application would require this feature?
Regards
Greg Cirino
Of Greg - Cirelle
Enterprises
Sent: Thursday, December 16, 2004 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] My Boss wants background music
At 04:48 AM 12/16/04, you wrote:
Dear Members,
I am searching for a new PBX for the company. My choice
At 09:01 PM 12/16/04, you wrote:
Paul Crick wrote:
But seriously, if you think you're owed karma for something and haven't
received it, flag it to a bug marshall. I'm not one, I just did the web
stuff.
funny thing that karma stuff, you are never owed any, you just keep doing
good stuff to
Has anybody seen this behaviour?
sip conf is stored in mysql database in 2 tables
ast_config for static (general) key/values
sip_buddies for sip extension detail.
database on the same machine as asterisk
Grandstream phones (I happen to have 2) register with asterisk
via sip with accounts and
At 07:05 AM 12/10/04, you wrote:
greg wrote:
Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel
Diego Aguirre wrote:
I use atendent transfer in Asterisk!!!
ok, the # extension key combo
too bad the 3 buttons cannot be programmed to emulate
the functions to make it work.
At 09:09 AM 12/10/04, you wrote:
On Friday 10 December 2004 13:53, Greg - Cirelle Enterprises wrote:
At 07:05 AM 12/10/04, you wrote:
greg wrote:
Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for
Nortel
Diego Aguirre wrote:
I use atendent transfer in Asterisk!!!
ok
At 09:59 AM 12/13/04, you wrote:
Get newest CVS. Its in there. Trust me. Oh..be sure your getting
asterisk-addons.
-Matthew
Got it thanks
Greg
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Just so I understand the data structure
and what goes in
Static configuration is where you can store regular *.conf files into the
database. These configurations are read at Asterisk startup/reload. Some
modules may also re-read this info upon their own reload (Ex. sip reload).
The table
At 06:14 PM 12/14/04, you wrote:
On December 14, 2004 04:11 pm, Greg - Cirelle Enterprises wrote:
It's been hours since I've seen a post from this list
Must be broken again.
So you'll email a broken list to send a message...? :-)
-A.
(yes I realize I'm replying to it)
: just making sure
At 11:02 AM 12/15/04, you wrote:
Your sip_buddies table should have 2 columns, allow and disallow. You
should be able to:
INSERT INTO sip_buddies (allow,disallow) VALUES (g729;g726;gsm,g711);
to give the equiv of:
allow=g729
allow=g726
allow=gsm
disallow=g711
-Matthew
I have the sip in 2 tables,
I park a call and instead of the parked extension
being returned, I get silence and the log shows
a bunch of the following messages
WARNING[26220]: codec_gsm.c:135 gsmtolin_framein: Huh?
A GSM frame that isn't a multiple of 33 or 65 bytes long from
(null) (320)?
what does this mean?
BTW these
At 09:38 AM 12/17/04, you wrote:
Hi,
Look in your sip.conf
host=192.168.20.2
and your phone is set to use 192.168.20.25
try to change host directive in sip.conf to
host=192.168.20.25
Diego Aguirre
Host is set to dynamic
host=dynamic
type=friend
I think this is an issue in the way chan_sip
I am trying to resolve a problem where grandstream phones (only)
fail to register after a period of time.
I have a mysql realtime setup that appears to work, but fails
for no reason.
before i classify the realtime system as unusable, I am trying
to isolate the problem.
One thing I have noticed is
having set up mysql per instructions for the voicemail system
in realtime, we have noticed, email notification has stopped
on receipt of voicemail.
this works fine on conf file setup, not under realtime.
Regards
Greg Cirino
___
At 09:46 AM 12/20/04, you wrote:
Actually, I got the display flashing when I have a new message. But it is
possible to get the Grandstream's Message button working? My goal is to
pickup earphone and press Message button to retrieve my messages.
Thanks.
update your firmware past 1.0.5.16 and put
Apparently asterisk cannot reboot gracefully (unattended)
when using realtime
MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1.
Check debug for more info.
WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug.
Since asterisk starts before mysql it
At 01:06 PM 12/20/04, you wrote:
Even if MySQL RealTime fails to connect to the database server on Asterisk
startup, Asterisk will continue to load. I just tested successfully tested
this.
MySQL RealTime will try to re-connect upon any further RealTime code
execution.
-Matthew
my installation of
At 10:57 AM 12/20/04, you wrote:
Have you looked at the debug logs to see if there is any SQL errors?
-Matthew
I realized the problem, was using a semi colon to separate addresses
to send the VM email notification from one extension to 2 addresses.
I solved that issue by using a forward address
At 03:50 PM 12/20/04, you wrote:
Matt Riddell wrote:
Sean Kennedy wrote:
Second thing is this: My office is scouting out VoIP solutions, and I
have suggested an asterisk solution. We will be getting our voice lines
over 9 channels of a t1. I feel comfortable enough with asterisk to set
this
At 04:42 PM 12/20/04, you wrote:
I went into the phone and made sure that 'User ID is phone number' was set
to 'No' and made sure that Fromuser=ext# was not present in the sip.conf
file. When a call comes in, the log shows the incoming number but the phone
still reads the extension number. I
Get out your wallets boys if you want to get sucked in by
this line
Microsoft did it and sucked in a bunch. that certification
and a 3.50 will get you a coffee at starbucks. IT jobs
are in the dumper anyway, so again why?
do you honestly think you are going to be asked for your
asterisk
At 04:17 PM 12/20/04, you wrote:
On December 20, 2004 04:02 pm, Greg - Cirelle Enterprises wrote:
Could I ask how you've connected the t1s? I'm going to be getting a
non-pri t1 ( 9 channels of voice, the rest off ). I assume I'll just
get an rj45(ish) plug to plug into the back of the card
maybe a dumb question but what do we have here???
sip-friends.sql
#
# Table structure for table `sipfriends`
#
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`username` varchar(40)
At 11:37 AM 12/21/04, you wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX account for each phone. I was EXPECTING them
to each
does anybody have an idea what the difference and significance
of sip seeding and registration is.
g
Regards
Greg Cirino
___
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603-425-2221
www.cirelle.com Web Application Development Design
www.cirelle.net ProSpeed High Speed Dial-up - 6
from the asterisk messages log:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25'
the only place I can see extension 40852 linked to the ip is in the
phone's configuration.
sip.conf
[40852]
;for a grandstream bt100
musicclass=homeline
pedantic=yes
accountcode = 40852
amaflags =
At 02:46 PM 12/21/04, you wrote:
Greg - Cirelle Enterprises wrote:
from the asterisk messages log:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25'
the only place I can see extension 40852 linked to the ip is in the
phone's configuration.
pedantic=yes
Take out pedantic=yes
At 03:37 PM 12/21/04, you wrote:
Hi again. I cant get my Budgetone registered in Asterisk, and I cant find
what's wrong... uff. This is my config:
This fragment is from my sip.conf:
[12345]
type=user
user=12345
username=12345
secret=12345
authuser=12345
qualify=1000
nat=no
host=dynamic
At 07:30 AM 12/22/04, you wrote:
I tried type=friend and it is registering now... I'm happy with it this
time, but why can't I have the phone as user only (only to make calls) and
not as peer (to receive calls)??
Thanks,
RODOLFO
Rodolfo Grave wrote:
Hi again. I cant get my Budgetone registered
At 09:44 AM 12/22/04, you wrote:
Its a way of storing ur sip stuff in a database rather than using the
flat files. Sip friends - extensions.conf stuff. Sip_buddies -
sip.conf stuff
___
this is the database to flat file storage I take it.
Regards
Greg
At 12:43 AM 12/22/04, you wrote:
Seeding occurs if there is still a persistent record (in astdb) of a preceding
location registration of a peer after a restart of asterisk or the sip
channel.
If Asterisk goes down and the peer has a long registration refresh time,
the phone maybe inaccessible
certify this
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At 12:21 PM 12/22/04, you wrote:
Did anyone here use the * forums over at asterisk.xvoip.com? I've been
unable to connect for a few days now and was wondering if anyone knew if
they're down for good.
It'd be a shame if they are since * newbs like me need every resource we can
find.
Joel Moore
If anyone is experiencing this type of registration error:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25'
try adding the following line to your modules.conf file
noload = app_adsiprog.so
This error is clearly asterisk trying to register with the phone
and not the other way
I know I'm getting tired of looking at code, but
why is the function register_verify defined in 2 different
files?
chan_iax2.c
line 3860
static int register_verify(int callno, struct sockaddr_in *sin, struct
iax_ies *ies)
chan_sip.c
line 4869
/*--- register_verify: Verify registration of user */
which docs are you talking about?
At 06:15 PM 12/22/04, you wrote:
Yeah, I d like to get those docs too.
--
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Wednesday, December 22, 2004 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
At 07:00 PM 12/22/04, you wrote:
What registration failure is that?
from the asterisk messages log:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25'
The only way to tell is a complete SIP trace of what's going on.
That may be, but the point is when the registration failure
Is there some reason the sipbuddies table structure was
designed with sip config values as column names?
Doesn't look very flexible
It really should take the form of ast_config so when
a new sip feature is implemented, you don't have to
re-write the entire data structure too.
Regards
Greg Cirino
Apparently, the realtime system in asterisk is faulty.
Implementing realtime, begins a host of seeding messages
along with registration messages visible at the CLI prompt.
This is not the case with .conf file configuration
Unfortunately, it is not clear where the bug originates
but is shows it's
At 09:53 AM 12/23/04, you wrote:
It was written the way it is because that is how RealTime works. =P If you
don't like the schema design, talk to Mark so he can rewrite RealTime for
you.
Read up some more on how RealTime works then you will understand why all the
tables are designed the way they
a column say musicclass = something,
or pedantic = yes like I could in the conf file and have it mean anything.
regards
greg
- Original Message -
From: Greg - Cirelle Enterprises [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
At 10:37 AM 12/23/04, you wrote:
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a
HOWTO somewhere?
At 10:43 AM 12/23/04, you wrote:
On December 23, 2004 10:37 am, Matthew Boehm wrote:
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
If Digium won't support it return the
At 11:17 AM 12/23/04, you wrote:
On December 23, 2004 10:59 am, Greg - Cirelle Enterprises wrote:
I spoke with a fellow, (can't remember his name, but had a british accent,
there
are only about 10 folks working there) at sangoma, and he specifically said
the
sangoma card will only work
At 11:52 AM 12/23/04, you wrote:
On December 23, 2004 11:14 am, TC wrote:
but thats the bitch Mark has put years of blood sweat into it,
now as asterisk start to become much bigger than the single developer/co
how do you divest
that control in a fair/equitable manner
I agree with you on all
At 12:14 PM 12/23/04, you wrote:
On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load
At 01:21 PM 12/23/04, you wrote:
Greg - Cirelle Enterprises wrote:
Read it, makes no difference, it's broken :)
Also, it doesn't say why the table structure is the
way it is. just poor data modeling.
God, I'm sure everyone on the list must be thinking, Oh, why oh why
didn't *Greg* write Asterisk
At 03:43 PM 12/23/04, you wrote:
Oh, I see. This is the realtime connected problem.
Can't say too much constructive about that without info, I'm not a fan of it.
We need a debug trace of the registration process (SIP trace and *
messages) to debug why it failed,
not just a one-line message, and
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