[top-posting continued]
Iconnecthere.com does indeed still work correctly with G711u and
G729, as tested by myself just a few moments ago.
JT
At 11:12 PM -0400 9/30/03, Andrew Joakimsen wrote:
When was this? I just completed a call with ICH and it was using ULAW.
While G723.1 support is possibl
I believe I"ve fixed the REGISTER channel leakage bug (it may fix other
potential channel leakages as well). Please confirm this fixes it. I do
*not* konw that this will affect general outgoing sip registry
reliability, more on that as I find it.
Mark
___
I was using the wrong table structure!! All I could find was one posted
on the list a while back and I didn't think much of it.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tilghman Lesher
> Sent: Tuesday, September 30, 2003 11:04
When was this? I just completed a call with ICH and it was using ULAW.
While G723.1 support is possible for asterisk, it is not standard and
ICH will work without it.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTE
On Tuesday 30 September 2003 21:55, Andrew Joakimsen wrote:
> The only thing I can think of is that my table/structure is
> incorrect, is there a sample somewhere?
Yes, see /usr/src/asterisk-addons/doc/cdr_mysql.txt
-Tilghman
___
Asterisk-Users mailing
The root issue is that ICH today stopped accepting any format other than
g.723.1 (which Asterisk doesn't support).
-alex
On Tue, 30 Sep 2003, Andrew Joakimsen wrote:
> Try
>
> exten => _71NXXNXX,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
>
> or
>
> exten => _7.,1,DIAL(SIP/${EXTEN:[EMAIL PROT
The only thing I can think of is that my table/structure is incorrect,
is there a sample somewhere?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tilghman Lesher
> Sent: Tuesday, September 30, 2003 10:28 PM
> To: [EMAIL PROTECTED]
Kevin:
try without the password to start with (remove it from the Budgetone).
reinvite=no.
Make sure you have disallow all codecs and allow ulaw/alaw.
In the Budgetone make sure you have the correct IP address to your Asterisk.
I would not use the defaultip.
In the Budgetone make sure DTMF via INFO
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
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 loggi
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
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
context=
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: [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 Grandstre
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 mail
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 license
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
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 aster
> 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 t
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 information
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 ast
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 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:
>
>
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 downl
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
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, t
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 J
You need an ADSI phone such as the nortel 350/390/480
Also, I believe you can only do this through a Zap interface, could be wrong
tho
Chad
>
> I don't have an adsi phone to test, but it should work?
___
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[EMAIL PROTECTED]
ht
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
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 t
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: [Asterisk-
Ok, I have 2 PC's that are identical. I loaded Redhat9 on the first and
Asterisk and installed a x100p card and it works fine. I ghost the first PC
and install the ghost on the second PC. Everything comes up on the 2nd PC
but I can't get it's x100p card to load. The systems sees the card, it's on
i
> -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)
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
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
Also,
http://npanxx.darkrealms.net/
http://members.dandy.net/~czg/
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
FYI to list.
Thx.
B.
--- Begin Message ---
Tilghman Lesher wrote:
Please post your entire voicemail.conf, with the general and
zonemessages sections, as well as your voicemailbox definition
(minus the password). Also, post the output of:
bash$ grep app_voicemail2.c /usr/src/asterisk/apps/CVS/Ent
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 is
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 ev
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 b
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:
>
>
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 conver
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 (
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
>-- 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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[EMAIL PROTECTED]
http://lists.
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
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 thes
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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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asteris
> -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 he
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: 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,
> > whic
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 => cdr_addo
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 adde
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
o
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
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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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 understandi
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 Se
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 un
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 r
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
> 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 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 conte
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 hav
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,
Steve
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
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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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 eq
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 sie
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
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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Whats the default SPEEX bitrate set to in Asterisk?
Later..
___
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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
>
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 p
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 stuf
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? Overtr
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
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 f
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
rj45
>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. reg
> 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 ring
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 l
> > 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
rj11<>bt
> > modem c
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 and also connect a normal UK
> phone respectively. I've seen a couple of people asking about it but not
> the answers.
>
> My question
> >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 rj11<>bt
> >modem cable
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
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 ma
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
RJ
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 i
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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> 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
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.
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
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 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
blah-b
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:
> 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 wa
>> 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 t
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