I wonder if I'm the only one who finds Allison reading the timestamp on
my voicemails in GMT, after months of having it done with the local time
I have set in voicemail.conf.
The timestamps on the message files, and the emails that are sent, are
correct. Only the spoken dates appear to be
Mark Evans wrote:
I think we're getting away from the original purpose of this program.
Are people really that desparate for a full, web-based admin/user
interface?
I sure am, I want to give as much control as I can for basic tasks
to my customer who may not even know what Linux is :)
On Tuesday 30 September 2003 01:01, Brian Capouch wrote:
I wonder if I'm the only one who finds Allison reading the timestamp
on my voicemails in GMT, after months of having it done with the
local time I have set in voicemail.conf.
The timestamps on the message files, and the emails that are
Jamie Carl wrote:
All registered with Sourceforge. The new project will be called
'AstWeb' and will be available under the GNU General Public Licence.
The current release of AstCDR will be ported over in the next day or
so along with the addition of a few features and labelled 'AstWeb v0.4'.
Brian West wrote:
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
I still don't get this.. Asterisk is GPL (with an option of a commercial
licence), MySQL is GPL(with an option of a commercial licence on the
Is GPL the correct licence for it??
I think GPL is the right licence, as long as people keep the code open
and share you rarely get licencing issues. In my experience it's the
ones who want to take it and then keep the changes hidden that cause the
problems and IMHO these people are in it for
To me it sounds like we have talked about the following..
* CDR Viewing (real time view of last 20 (filterable by
src/dst/accountcode etc..) CDR entries)
* CDR Reporting (per user/company/line, Somthing like the
itemised billing from a telco, User Accessable)
Sounds good. Would you want
Personally, from my experiences with integrating different open source
libraries, the BSD license make good sense (hey, if its good enough for
Postgresql, its fine for me :)
-Bryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Evans
Sent:
On Tue, 30 Sep 2003 07:43:34 +0100
WipeOut [EMAIL PROTECTED] wrote:
Thats COOL!!!. :)
Is GPL the correct licence for it??
I am not so hot on all that licencing stuff and all I
hear is that licence X is not compatible with licence Y
and so this code needs to removed and yada-yada-yada
Mark Evans wrote:
Is GPL the correct licence for it??
I think GPL is the right licence, as long as people keep the code open
and share you rarely get licencing issues. In my experience it's the
ones who want to take it and then keep the changes hidden that cause the
problems and IMHO these
On Tue, 30 Sep 2003 00:02:06 -0700
Paul Crick [EMAIL PROTECTED] wrote:
itemised billing from a telco, User Accessable)
Sounds good. Would you want to extend it any, using place
names etc? So
calls to +1604xxx show up as Canada or BC, Canada (or
even Vancouver
BC, Canada) or am I just being a
Paul Crick wrote:
To me it sounds like we have talked about the following..
* CDR Viewing (real time view of last 20 (filterable by
src/dst/accountcode etc..) CDR entries)
* CDR Reporting (per user/company/line, Somthing like the
itemised billing from a telco, User Accessable)
Sounds good.
The problem with this is that its a major problem mapping
area codes of the various contries of the world.. (especially
Canada where I have been told that landline and cell phone
networks use the same codes.. )
Yup, it's true - I know so cos I'm in Canada :-) (and coming from the UK,
don't
hi all
after setting up chan_h323, I don't get any ring indications on my Dlink
DPH-100H phone. Any idea how to debug this?
roy
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Paul Crick wrote:
I meant using the PHPconfig stuff as is.. it edits the
.conf files directly which to me seems the best idea at
this point..
Gotcha.. so stick with the text area editing a text .conf file thing then -
sounds cool.. then have a link off to the CDR bit etc?
Or integrate it
Hi,
I've found this thread in the archives and am currently planning to
connect my digium kit to a BT landline and also connect a normal UK
phone respectively. I've seen a couple of people asking about it but not
the answers.
My question is - do I need a modtap (listed at maplin.co.uk as BT to
Matthew J Keay wrote:
Hi,
I've found this thread in the archives and am currently planning to
connect my digium kit to a BT landline and also connect a normal UK
phone respectively. I've seen a couple of people asking about it but not
the answers.
My question is - do I need a modtap (listed at
The TDM400p is a fxs device, meaning you cannot connect it to a line socket,
you can only connect it to a handset. to connect your * box to BT you will
need x100p cards.
for the handset a straighthrough rj11 should work.
On Tuesday 30 September 2003 11:51 am, Matthew J Keay wrote:
Hi,
I've
My question is - do I need a modtap (listed at maplin.co.uk as BT to
RJ45 adapters) full master with line protection, pabx master without
line protection or a pabx slave socket. --or-- can I just use a
straight-thru rj11 cable for the handset and a straight-thru
rj11bt
modem cable for
Sorry, I was confused, I also have an X100P, I should read the subject.
Michael
On Tue, 30 Sep 2003, Michael T Farnworth wrote:
On Tue, 30 Sep 2003, Matthew J Keay wrote:
Hi,
I've found this thread in the archives and am currently planning to
connect my digium kit to a BT landline
Sorry, I was confused, I also have an X100P, I should read the
subject.
Michael
I used a simple converter for BT to RJ45 that came with an old
modem, it
seems to work fine. I tried plugging in a handset cable and it
didn't
work, in fact phones elsewhere in the house started ringing in a
I like the ideas of user self service.. or user filtered
access to the CDR stuff?
Will there/should there be levels of user access and
stuff to restrict access to certain areas?
From an architectural viewpoint, there will likely be three drivers
for some form of restricted access:
a.
Only 2 wires needed, 2 center wires from the rj11 to 2nd and 5th on the
bt plug.
This can be used to plug in the x100p to a BT socket. If your telco
provides rj45 termination (pbx etc), a straight through rj45 works fine.
The TDM terminates in rj45 (psudo PBX), you can plug a straight through
Hi all,
I've got my dirty hands on (free!) Vocaltec 4-port FXO/FXS gateway. It is
used unit, I managed to configure correct IP settings in it but am somewhat
at loss how to integrate it into my existing Asterisk network. I have no
H323 gatekeeper, no Vocaltec Network Manager software, and am not
According to Troy Settle:
Why do they do that? Quite possibly because they, like myself, hate
having to scroll through pages and pages of quotes to get to the reply,
which isn't always clear where it might start.
Troy, you're not complaining about bottom-posting; you're complaining about
On Tue, 30 Sep 2003, Roderick Montgomery wrote:
According to Troy Settle:
Why do they do that? Quite possibly because they, like myself, hate
having to scroll through pages and pages of quotes to get to the reply,
which isn't always clear where it might start.
Do what? Overtrimming
On Mon, 29 Sep 2003, Brad Bergman wrote:
The M phones from Nortel are digital phones as used with Norstar or
Meridian 1 systems.
Actually some if not all M8XX and M9XX phones, the 9516 for example (which are
now sold by Aastra) are just analog phones with lots of buttons and stuff...
See my Mon, 29 Sep 2003 10:30:23 -0400 email (Sorry emails have no message #s to refer
to :) )
-- Original Message --
From: Keith O'Brien [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Tue, 30 Sep 2003 00:25:30 -0400
I think that you missed my
On Tue, 30 Sep 2003, Keith O'Brien wrote:
I think that you missed my point. I am not proposing to establish a forum
and abolish the maillist.
The forum would get traffic as all posts sent to the maillist would
automatically post to the forum. Those that want to answer using the forum
can
Whats the default SPEEX bitrate set to in Asterisk?
Later..
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Hello,
Someone experienced about using application Flash?
Would you please be kind and send some working examples of extensions
using Flash?
-Johanna
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On Tue, 2003-09-30 at 07:53, costas wrote:
See my Mon, 29 Sep 2003 10:30:23 -0400 email (Sorry emails have no
message #s to refer to :) )
This is why top posting bites. What the hell are you talking about?
-- Original Message --
From: Keith O'Brien
On Tue, 30 Sep 2003, Johanna Kangas wrote:
Someone experienced about using application Flash?
Would you please be kind and send some working examples of extensions
using Flash?
you can only use Flash on zaptel devices, we had it done coz i wanted to
be able to dial a number onto a dumb
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
To try and sort out the problem I tried to register to Sipcall with
Linphone and sent the dialogs to tech support of the
Just to be perfectly clear here. You *never* plug two FXS devices
together unless you for some reason do not wish them to ever work again.
Mark
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Comments inline:
On Tue, Sep 30, 2003 at 07:38:07AM +0100, WipeOut wrote:
Mark Evans wrote:
I think we're getting away from the original purpose of this program.
Are people really that desparate for a full, web-based admin/user
interface?
I sure am, I want to give as much
If shorting two FXS lines together damages them they are badly designed.
Good BORSCHT (battery, over-voltage protection, ringing, signaling,
hybrid, and test) design should mean they can tolerate this kind of
thing. They have to very often in the poorly controlled PSTN rats nest.
Regards,
PJ Welsh wrote:
Comments inline:
Do you want everyone to know where everyone is calling? I could see mgmt/owners having an issue with this in a normal business environment! Look Mr. Gate$ at extention 666 made 500 calls to Mr Devil last week, I wonder what deals they are making ;)
This would
I can add stuff to the phpconfig CVS dirs if anyone wants to contrib stuff.keep it
all in one dir etc
phpconfig can run SSL and yes its a easy way to vi the files directly but it does
give your options that vi does not like click on a context and see extensions
within that context.
If shorting two FXS lines together damages them they are badly designed.
Good BORSCHT (battery, over-voltage protection, ringing, signaling,
hybrid, and test) design should mean they can tolerate this kind of
thing. They have to very often in the poorly controlled PSTN rats nest.
Generally
I understand Asterisk have contributions of many
developers and not only from Digium.
When licensing Asterisk from Digium looking other kind
of licence than GPL, what happens with the rights of
other members of this community?
Alejandro
--- Uriel Carrasquilla [EMAIL PROTECTED] escreveu:
If I
I'm running * on two pstn lines (x100p cards) that happen to also have
analog phones installed on the incoming pair for backup (until testing is
complete).
If someone is talking on a pstn analog phone and they talk loud
enough, * senses the voice (apparently assuming the line is ringing),
and
Alejandro Olchik wrote:
I understand Asterisk have contributions of many
developers and not only from Digium.
When licensing Asterisk from Digium looking other kind
of licence than GPL, what happens with the rights of
other members of this community?
Alejandro
No code is commited to Asterisk
No GPL software can touch asterisk. If it does then asterisk would be
encumbered by the GPL and the dual lic. model digium has would be shot.
They would no longer have the abililty to lic. a comercial version of
asterisk outside of the GPL.
bkw
PS or atleast thats my understanding.
On Tue, 30
Brian West wrote:
No GPL software can touch asterisk. If it does then asterisk would be
encumbered by the GPL and the dual lic. model digium has would be shot.
They would no longer have the abililty to lic. a comercial version of
asterisk outside of the GPL.
bkw
PS or atleast thats my
Brian West wrote:
http://www.loligo.com/asterisk/current/
Hey, didn't think of that. Good idea. Will go and check it out.
Thanks,
Leif Madsen.
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On Tue, 2003-09-30 at 09:27, WipeOut wrote:
Alejandro Olchik wrote:
I understand Asterisk have contributions of many
developers and not only from Digium.
When licensing Asterisk from Digium looking other kind
of licence than GPL, what happens with the rights of
other members of this
Steven Critchfield wrote:
On Tue, 2003-09-30 at 09:27, WipeOut wrote:
Alejandro Olchik wrote:
I understand Asterisk have contributions of many
developers and not only from Digium.
When licensing Asterisk from Digium looking other kind
of licence than GPL, what happens with the rights of
Hi all,
I think I've run out of options in terms of what I know about this.
I have created a user called asteriskuser and granted all privileges to
the asteriskcdrdb database. Then I created the table via the
cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect
this, and
On Tuesday 30 September 2003 10:57, Leif Madsen wrote:
I have created a user called asteriskuser and granted all privileges
to the asteriskcdrdb database. Then I created the table via the
cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect
this, and added load =
-Original Message-
From: Roderick Montgomery
Sent: Tuesday, September 30, 2003 8:24 AM
According to Troy Settle:
Why do they do that? Quite possibly because they, like myself, hate
having to scroll through pages and pages of quotes to get
to the reply,
which isn't
Is there a trick to making ADSI work in a T100P-Channel Bank
environment? Or is (as I suspect) ADSI simply not supported by my CSC
Access Bank II hardware?
-- Executing ADSIProg(Zap/1-1, asterisk.adsi) in new stack
-- ADSI Unavailable on CPE. Not bothering to try.
Mark
-Original Message-
From: Steven Critchfield
Sent: Tuesday, September 30, 2003 9:23 AM
On Tue, 2003-09-30 at 07:53, costas wrote:
See my Mon, 29 Sep 2003 10:30:23 -0400 email (Sorry emails have no
message #s to refer to :) )
This is why top posting bites. What the hell are you
Tilghman Lesher wrote:
Try this command (from your bash shell):
grep cdr_mysql /var/log/asterisk/messages
messages does not exist... ?
Thanks,
Leif Madsen.
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On Tue, 30 Sep 2003, WipeOut wrote:
Whats the default SPEEX bitrate set to in Asterisk?
The default bitrate for speex (at this time determined by the speex lib
because we don't explicitly set it) is 15k
I'm still looking for a good way to implement options for codecs so we can
modify these
On Tuesday 30 September 2003 11:52, Leif Madsen wrote:
Tilghman Lesher wrote:
Try this command (from your bash shell):
grep cdr_mysql /var/log/asterisk/messages
messages does not exist... ?
Try uncommenting the line in /etc/asterisk/logger.conf. Restart your
Asterisk process, try a few
-- Executing ADSIProg(Zap/1-1, asterisk.adsi) in new stack
-- ADSI Unavailable on CPE. Not bothering to try.
do you have adsi=yes zapata.conf b4 the channels that have the
adsi phones
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Make sure you set adsi=yes in your zapata.conf, and that you are using an
adsi compatible device.
Chad
- Original Message -
From: Mark Farver [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 30, 2003 12:38 PM
Subject: [Asterisk-Users] ADSI only works with what?
Is there
Have a way to specify it in the src? I would like to try the 8k between a
few servers and see how it sounds.
bkw
On Tue, 30 Sep 2003, James Golovich wrote:
On Tue, 30 Sep 2003, WipeOut wrote:
Whats the default SPEEX bitrate set to in Asterisk?
The default bitrate for speex (at this
Tilghman Lesher wrote:
Try uncommenting the line in /etc/asterisk/logger.conf. Restart your
Asterisk process, try a few calls and do the above command again.
Aha! You've helped me solve a problem which I didn't realize existed :)
In an attempt to try to get into the habit of following up
On Tuesday 30 September 2003 12:43, Leif Madsen wrote:
logger.conf was incorrectly setup, for some reason in the moving of
files around on my system, my logger.conf file was over-written with
manager.conf, hence why I was not getting any logging. Fixed that by
using this configuration:
;
On Tue, 30 Sep 2003 00:02:06 -0700
Paul Crick [EMAIL PROTECTED] wrote:
itemised billing from a telco, User Accessable)
Sounds good. Would you want to extend it any, using place names etc? So
calls to +1604xxx show up as Canada or BC, Canada (or even Vancouver
BC, Canada) or am I just being a
Dave Cotton wrote:
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
Does this imply that it will work even in a NAT environment?
I have watched the list like a hawk for
Hello,
I have my system up and running just fine with 2 voice T1s(both B8ZS non-PRI
non-ISDN) They work fine for outgoing and incoming calls. I wanted to get
callerID detection working but I cannot. I am receiving the 4-digit DNIS as
the extension variable in Asterisk, but the callerid variable
Hi!
I have that message:
*CLI WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 177 (Request)
I was thinking..why that call is for 127.0.0.1 is it the loopback of the
asterisk machine?
Thanks for any help
Miklos
Hi!
I have a strange problem with ICH calls.
When i try to make a call with asterisk for ICH nothing happens ( register
is ok)
But when i register my snom 200 with ich it works very well with the same
register data.
Someone knows anything about?
miklos
-Original Message-
From: Jamie Carl
Sent: Monday, September 29, 2003 7:44 PM
Guys! I'm putting the source up on SourceForge on my
existing account. Questions is this tho:
Suggestions please! I would like to get this on SF by the
end of the day. (it's 9:33am here).
Please post your extensions.conf and sip.conf sections relevant to
ich/deltathree.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 30, 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject:
I was told ADSI would not work on a dlink gateway, after setting
adsi=yea in mgcp.conf I now get:
Executing ADSIProg(MGCP/aaln/[EMAIL PROTECTED], ) in new stack
-- ADSI Available on CPE. Attempting Upload.
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'D'
I don't have an adsi phone to
Got 3 core dumps from asterisk in a very short period of time, not sure
what was going on, but I am posting these in case anybody wants to see
them, if you want to see more info I can provide it on request.
I am using the asterisk 0.5.0 download.
(gdb) bt
#0 ast_translator_free_path (p=0x10) at
Ok
extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
sip.conf:
register =31451543:[EMAIL PROTECTED]/33
[iconnect]
type=friend
secret=
username=31451543
host=sipauth.deltathree.com
dtmfmode=inband
context=from-sip
miklos
- Original Message -
From: Andrew
On Tue, 2003-09-30 at 20:21, Brian Capouch wrote:
Does this imply that it will work even in a NAT environment?
I have watched the list like a hawk for evidence of FWD working for
machines placed behind NAT, but so far haven't seen that anyone could
actually get it going.
If so, that
Try
exten = _71NXXNXX,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
or
exten = _7.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
Regards,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 30, 2003 4:52
cvs update
Jeremy McNamara
Michael T Farnworth wrote:
Got 3 core dumps from asterisk in a very short period of time, not sure
what was going on, but I am posting these in case anybody wants to see
them, if you want to see more info I can provide it on request.
I am using the asterisk 0.5.0
[snip happens]
According to John Todd:
On Tue, 30 Sep 2003 00:02:06 -0700
Paul Crick [EMAIL PROTECTED] wrote:
oh my god, who's going to collate a list of Names/Area-codes?? Stuffed
if I'm doing it. :)
Or, at least one of these sources might have the data for free:
(harvested off
cvs update
Got 3 core dumps from asterisk in a very short period of time, not sure
what was going on, but I am posting these in case anybody wants to see
them, if you want to see more info I can provide it on request.
I am using the asterisk 0.5.0 download.
he mentioned he was using the asterisk
On Tue, 30 Sep 2003, duncan wrote:
cvs update
Got 3 core dumps from asterisk in a very short period of time, not sure
what was going on, but I am posting these in case anybody wants to see
them, if you want to see more info I can provide it on request.
I am using the asterisk 0.5.0
Tilghman Lesher wrote:
I would recommend that you remove debug from logging, unless you have
a situation where you really need it. The debug log contains a lot of
information, which can consume disk space quite quickly. In addition,
because of the verbosity, the debug log can contain
he mentioned he was using the asterisk 0.5.0 download though. surely this
means we should update the 0.5.0 release to solve these problems?
can you confirm that it was the asterisk-0.5.0.tar.gz file install that
caused these segfaults?
It was, I just downloaded the tar file, I haven't
We should be prtty oeto a 0.5.1 in the net few days.
Mark
On Wed, 1 Oct 2003, duncan wrote:
he mentioned he was using the asterisk 0.5.0 download though. surely this
means we should update the 0.5.0 release to solve these problems?
can you confirm that it was the
How did you get it to work? I cannot figure out how to get mysql cdrs
working, all I get is:
ERROR[16401]: File cdr_mysql.c, Line 130 (mysql_log): Failed to insert
into database.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leif
Troy Settle wrote:
With all the discussion about licensing issues and the sort, I think it's
time for a full blown 3rd party application to work with Asterisk while at
the same time not causing Asterisk to become encumbered. For such a
project, I'm license neutral. While I prefer the BSD
When I dial with my Grandstream 101 telephone to another sip phone or
Zap FXS, the call rings, but no audio is passed. Eventually the call
gets disconnected. The same thing happens if I dial the Grandstream.
Any Suggestions?
___
Asterisk-Users
Any nat involved? and what codec's are you trying?
On Tue, 30 Sep 2003, Kevin wrote:
When I dial with my Grandstream 101 telephone to another sip phone or
Zap FXS, the call rings, but no audio is passed. Eventually the call
gets disconnected. The same thing happens if I dial the
Are the SIP phones behind a NAT/Router? is * behind a NAT or Firewall?
can we see your sip.conf file?
Regards,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin
Sent: Tuesday, September 30, 2003 8:05 PM
To: [EMAIL PROTECTED]
Subject:
Thanks for the help,
Sorry about not originally providing that information. It's a 10. Local
area network no Nat involved. I am using the default setting of the
Grandstream and the following sip.conf
[gstream]
type=friend
username=gstream
secret=test
host=dynamic
defaultip=192.168.0.7
On Tue, 30 Sep 2003, Mark Spencer wrote:
We should be prtty oeto a 0.5.1 in the net few days.
Once Mark puts down the bottle, that is :-)
On Wed, 1 Oct 2003, duncan wrote:
he mentioned he was using the asterisk 0.5.0 download though. surely this
means we should update the 0.5.0
On Tuesday 30 September 2003 17:48, Andrew Joakimsen wrote:
How did you get it to work? I cannot figure out how to get mysql cdrs
working, all I get is:
ERROR[16401]: File cdr_mysql.c, Line 130 (mysql_log): Failed to
insert into database.
Well, there's many possible reasons why the logging
I've just received a brand new zhone z-plex 10b. At first, I was able to
login using the default (admin/zhone), and started exploring with
the configuration, trying to set it up to work with the TE410P.
Somehow the admin password was changed, but I don't
know what it is anymore. So now I'm stuck
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