Try deleting the line
pritrustusercid=yes
in zapata.conf
maxx
- Original Message -
From: "Bastian Schern" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 8:34 PM
Subject: [Asterisk-Users] Incoming MSN via ZapHFC -> to SIP
> Hi there,
>
> I've got a small prob
Have a look at http://www.voip-info.org/tiki-index.php?page=Cisco%20POE
-Shaun
- Original Message -
From: Gonzalo Gasca Meza <[EMAIL PROTECTED]>
Date: Fri, 20 Aug 2004 21:57:01 -0700 (PDT)
Subject: [Asterisk-Users] PoE injectors
To: [EMAIL PROTECTED]
Anyone knows some home-use PoE injec
Anyone knows some home-use PoE injector that works ok with Cisco 7960s?
Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.
Craig Guy wrote:
cancallforward=yes
There is no such function in distributed chan_sip.c,
ergo there can't be such a configuration parameter.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNS
Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html
Two new products
* A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter
* A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router
Jim
James H. Thompson
[EMAIL PROTECTED]
___
As described, you would need two FXO ports and two PSTN lines.
Alternatively, you could use one FXO port, one PSTN line, and an
outbound connection to a VOIP provider using SIP or IAX, if you have
network bandwidth available for that. Your setup and fixed recurring
costs would be lower, but whethe
Hi Will,
That won't work (I tried it originally). What happens is that an invalid
extension will match the _123495XX pattern and asterisk will try to dial it.
My understanding is that the i extension will only activate if no pattern
match can be found, eg if somehow 9[0-4,6-9]XX gets into the con
At 1:18 PM +0400 on 8/12/04, Lev wrote:
Hello.
I have not seen that asterisk
software have a possibility
to dial pulse on outgoing calls.
Don't you know is there any
plan to do it?
Thanks.
Good luck.
Lev.
There are patches in the bugtracker going in this direction...
http://bugs.digium.com/bug_view
At 5:07 PM -0600 on 8/20/04, Luke Cyca wrote:
Hello,
I'd like to be able to see if a channel is use and handle the call
differently if it is. The best I can find is the command
ChanIsAvail(). The problem is, I have an snom200 phone which does
call waiting, so even if it is engaged in a call, a
Hmm,
Asterisk is certainly recognising the call forward. I had this error when I
first tried it and resolved by adding the 'cancallforward=yes' in the
sip.conf for that extension. I think you may actually have to restart
asterisk to enable these functions rather than reload? I am also running
t
Hi there. I've got a question about what I think is a non-standard
usage for asterisk. What I want to do is receive inbound calls on an
analog line and connect that call to another number out there on the
PSTN (actually, I want to alternate calls between two numbers, but
that's not the part I
Kevin Day wrote:
I've got a Cisco 7960 that I'm trying to convert to SIP. Here's what
"Firmware Versions" says:
App Load ID:
P00305000300
Boot Load ID:
PC03M030
Version:
5.0(3.0)
The files that I have are:
P003-07-1-00.sbn
P003-07-1-00.bin
P0S3-07-1-00.loads
P0S3-07-1-00.sb2
If I put "P003-07-
Chris Shaw wrote:
If you really want to be able to telnet in as root, locate
telnetd.conf or somesuch and it should be in there somewhere
as a yes/no. (It is for ssh anyway..)
No, not under any distro I'm familiar with... It's under /etc/securetty...
You add the tty of the device you want to
On Fri, Aug 20, 2004 at 07:14:06PM -0400, Michael George wrote:
> I'm trying to add an emergency dial to my context. However, when I try to
> dial it, I get caught in an endless loop.
>
> For debugging, I have pared out nearly all the control flow and just have
> ChanIsAvail() and Dial() called.
I forgot to mention, these are SIP calls, I use a Pure SIP configuration...
it's so strange, if I remove macros and do the EXACT same thing without
macros, it works perfectly!
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.dig
Ok.. this is really wierd... I just cleaned up my dialplan a bit by adding
some macros with a strange side effect...
On my incoming context which has no macros in it, far end ringing used to
work... now that I have macros defined, far end ringing has stopped working
all together...
The macros DO
http://www.voip-info.org/wiki-asterisk+pbx+functions
http://www.voip-info.org/wiki-asterisk+vertical+service+activation+codes
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 3:02 PM
Subject: Re: [Asterisk-Users] Asterisk
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two differe
Hello,
I'd like to be able to see if a channel is use and handle the call
differently if it is. The best I can find is the command
ChanIsAvail(). The problem is, I have an snom200 phone which does call
waiting, so even if it is engaged in a call, a second channel is still
available on it. I
Hi,
sorry for interruption, but are there any guides for all possible Asterisk
PBX functions that are available with no particular dialplan handling ?
Thanks,
Robert.
- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 6:0
Hi All
BT are providing a SIP gateway for PSTN through the BT communicator with
Yahoo Messenger, I have done an ethereal trace and found that the BT
Communicator side of the software is using SIP, so in theory I could add
more PSTN lines to Asterisk for BT using SIP, but I am having problems
d
Imran,
The 'wcfxs' is for the TDM400P card and the 'wcfxo' is for the X100P card.
I found that to be quite confusing at first also.
Lyle
- Original Message -
From: "Imran Akbar" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 5:35 AM
Subject: [Asterisk-Users] x
I've seen other request info on this config before.
I've got the system up and running, Zap/Modem has been working for
several days, no errors/warnings.
Simply added info for broadvoice and I can dial into my system via the
remote, but dial out from my box renders the following:
I get one ring o
I had this exact issue. The 7.1 firmware has some issue where it
won't upgrade a SCCP image. I had to upgrade to phone to SIP 6.3 then
to 7.1
That was the key for me.
-Matt
On Fri, 20 Aug 2004 17:49:42 -0400, Doug Shubert <[EMAIL PROTECTED]> wrote:
> what version of SCCP are you running?
> Cis
Hi all
I have question regarding to my nom 200 and asterisk.
I have an * server with two x101p and two lines conected.
When i am in a call in line 1 and a call in line two is received the first
call goes imediatly to hold and the line button blinks indicating that
another call arrived.
It is ve
what version of SCCP are you running?
Cisco support link for converting SCCP to SIP
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#topic2
Kevin Day wrote:
On Aug 20, 2004, at 3:21 PM, Jerimiah Cole wrote:
Kevin Day wrote:
If I put "P003-07-1-00" in OS79XX.T
Scott Laird wrote:
I should probably mention that XO will sell "integrated PRIs", with 1 D
channel, N B channels, and all of the remaining channels used as a
channelized Internet-access T1. They quoted me $550/month last month in
Seattle. We're not taking them up on it for various reasons, but
> On Aug 20, 2004, at 3:21 PM, Jerimiah Cole wrote:
>
> > Kevin Day wrote:
> >> If I put "P003-07-1-00" in OS79XX.TXT, the phone tries to tftp
> >> XMLDefault.cnf.xml. I've tried every imaginable
> >> parameter, and can't get the phone to actually grab the image, it
> >> just keeps redownloadi
You need the following entries:
OS79XX.TXT
P003-07-1-00
SIPDefault.cnf
image_version: P0S3-07-1-00
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Day
Sent: Saturday, 21 August 2004 6:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I just put together a SuSE 9.1 box and got the * and zaptel drivers today.
Everything built fine but now I get:
FATAL: Error inserting zaptel (/lib/modules/2.6.5-7.104-smp/misc/zaptel.ko):
Invalid module format
Can't find too much in the list on it for this, just ztdummy. First thing
I'll try is
On Fri, 2004-08-20 at 15:47, Chris Shaw wrote:
> True, very true... If it's PRI then you will get DNIS/DNID from the
> D-Channel...
>
> If they're doing anything other than PRI though, like a regular T1 into a
> channel bank (or into a TE100P or TE40xP) , this would work... Lines would
> be assign
Ok, i haven't get in hand the card and make remote hardware install.
It's certainely the problem.
Thanks !
At 22:49 20/08/2004, you wrote:
On Fri, 20 Aug 2004, Arnaud Pignard wrote:
> I try setup a TE410P. Already setup E100P without problem. I also check
> sample zaptel.conf config in mailing list
oops. sent this last night - I was getting this list on an alias that I didn't
send from.
Hello,
I am attempting to setup spandsp on debian-unstable-2.6.7-recent-HEAD. Gnome
requires libtiff3g 3.6.1-1. I downloaded libtiff 3.5.7 and 3.6.0 and was
able to install 3.5.7,spandsp, and reinstall as
> My first and only unit cooked itself. It literally melted the casing.
> Sipura replaced it very promptly though.
Wow that's bad! Bad power supply?
Hmmm I don't think he meant 'flame' literally... lol...
___
Asterisk-Users mailing list
[EMAIL PROTECT
Hi,
I try to connect with -h and is working. Now I will try to create other db
and I will see... :(
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
McClintock
Sent: Friday, August 20, 2004 4:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [
My first and only unit cooked itself. It literally melted the casing.
Sipura replaced it very promptly though.
-Original Message-
From: Matt Schulte [mailto:[EMAIL PROTECTED]
Sent: Friday, August 20, 2004 7:44 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura endpoints
Anyon
Except for what you said about 1-800 numbers pointing to the same line...
nevermind I'll shut up now...
-Chris
- Original Message -
From: "Chris Shaw" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 1:47 PM
Subject: Re: [Asterisk-Users] determining what numbe
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 1:37 PM
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
> Chris Shaw wrote:
>
> >>I am suprised that one would have to create a dialplan since its an
On Fri, 20 Aug 2004, Arnaud Pignard wrote:
> I try setup a TE410P. Already setup E100P without problem. I also check
> sample zaptel.conf config in mailing list and seems my config is ok.
> However when i modprobe wct4xxp, here is error output :
>
> ZT_CHANCONFIG failed on channel 97: No such de
True, very true... If it's PRI then you will get DNIS/DNID from the
D-Channel...
If they're doing anything other than PRI though, like a regular T1 into a
channel bank (or into a TE100P or TE40xP) , this would work... Lines would
be assigned to a specific channel and they could be separated out wi
On Aug 20, 2004, at 3:21 PM, Jerimiah Cole wrote:
Kevin Day wrote:
If I put "P003-07-1-00" in OS79XX.TXT, the phone tries to tftp
XMLDefault.cnf.xml. I've tried every imaginable
parameter, and can't get the phone to actually grab the image, it
just keeps redownloading the XML file.
If I put "P0
Chris Shaw wrote:
I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?
I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service
On Fri, 2004-08-20 at 15:23, Chris Shaw wrote:
> Why not use separate contexts for these lines in zapata.conf? Seems way
> simpler to me...
>
> http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
Who said they are seperate lines. 1-800 numbers can just be redirects to
other lines. In that c
Hi,
I try setup a TE410P. Already setup E100P without problem. I also check
sample zaptel.conf config in mailing list and seems my config is ok.
However when i modprobe wct4xxp, here is error output :
ZT_CHANCONFIG failed on channel 97: No such device or address (6)
FATAL: Error running install c
Still waiting on Polycom for something. Will make it available as soon
as I get it.
John
Matt Darnell wrote:
That was part of my problem.
I can now get the 600 to download XML, I tried using
http://phone-xml.berbee.com/menu.xml and the phone displays "XML Error
(1,0) syntax error". I'm guessing
Can you connect to mysql from the command line with the user/pass you
setup?
Also, when you test make sure you add: -h localhost to the flags you
pass 'mysql'. This will make sure it doesn't try connecting via the
unix domain socket. The permissions in the 'mysql' db may be set not to
allow loca
Why not use separate contexts for these lines in zapata.conf? Seems way
simpler to me...
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
-Chris
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 12:33 PM
Sub
Kevin Day wrote:
If I put "P003-07-1-00" in OS79XX.TXT, the phone tries to tftp
XMLDefault.cnf.xml. I've tried every imaginable
parameter, and can't get the phone to actually grab the image, it just
keeps redownloading the XML file.
If I put "P0S3-07-1-00" in OS79XX.TXT, the phone wants to dow
Hi all:
Is it possible to setup the TE405P as follow for PRI E1 ?
zaptel
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
and the second span as well as
span=2,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
The thing is, I am trying to connect * to Nortel DMS100 MSC which does
Hi,
The same error:((
Aug 20 16:13:39 ERROR[278544]: cdr_addon_mysql.c:378 my_load_module: Failed
to connect to mysql database asteriskcdrdb on localhost.
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcolm Bader
Sent:
Did you create a user for the asteriskcdrbd?
In my case I created a user asteriskuser with all priviages and gave it
a password.
Then entered that info, user name and password, into
/etc/asterisk/cdr-mysql.conf
Malcolm
Chris HARIGA wrote:
Hi,
I have Fedora Core 2 running with a T1 card. I try to
I've got a Cisco 7960 that I'm trying to convert to SIP. Here's what
"Firmware Versions" says:
App Load ID:
P00305000300
Boot Load ID:
PC03M030
Version:
5.0(3.0)
The files that I have are:
P003-07-1-00.sbn
P003-07-1-00.bin
P0S3-07-1-00.loads
P0S3-07-1-00.sb2
If I put "P003-07-1-00" in OS79XX.T
On Fri, 2004-08-20 at 14:38, Nick Cobley wrote:
> I am currently recording incoming calls using the monitor application.
> Problem is it is recording silence for the ringing portion of the SIP
> call. We really would like it to commence from the beginning of the
> conversation. Is there any way
On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote:
> Hey all,
>
> I've setup * to serve the needs of our small helpdesk and I'm looking
> to expand. We're planning on doing support for different companies,
> each one identified by a different 1-800 number that terminates at our
> PBX. What I woul
I am currently recording incoming calls using the monitor application.
Problem is it is recording silence for the ringing portion of the SIP
call. We really would like it to commence from the beginning of the
conversation. Is there any way to do this?
Also, I tried setting monitor to record the
Hello Paul,
Check out this varialbe:
${DNID}: Dialed Number Identifier
Here are some others:
http://www.voip-info.org/wiki-Asterisk+Variables
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beh
Hey all,
I've setup * to serve the needs of our small helpdesk and I'm looking
to expand. We're planning on doing support for different companies,
each one identified by a different 1-800 number that terminates at our
PBX. What I would like to know is: is there a variable I can read to
determine w
Hi,
I have Fedora Core 2 running with a T1 card. I try to put
the log on db but I get the error:
Aug 20 15:17:47
ERROR[262160]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql
database asteriskcdrdb on localhost.
The database exists and I try with “mysqlaccess
l
Hi,
I have the following topology:
PSTN/H323 gateway->GNUGK->chan_h323/chan_sip->SIP EP
Mostly everything works fine except chan_h323 is not passing
audio from PSTN before the call is answered and as a result users
can't hear PSTN announcements (like "the number is not in service")
that's played
On Thu, 19 Aug 2004, Gary Pigott wrote:
| We're almost there the next problem is with the inbound calls over i4l.
|
| When the call is received, the "thank you for calling. press 1 for"
| announcement is supposed to start... but something strange happens. The last
| fraction of a second o
We have about 8 SPA-2000's and one SPA-3000 right now, and going to be
ordering about 10 more in the near future. We had one unit that was
"Dead on Arrival", it wouldn't make an ethernet connection, but it
turned out to be just an issue with the pins in the jack on the SPA2K
weren't high en
Hi there,
I've got a small problem with the zaphfc channel. No MSN of an any
incoming call which comes trough the ISDN card (Acer ISDN, with HFC
chipset and zaphfc driver) which will be forwarded to the SIP-Phone will
be displayed. Always it will be shown "asterisk" an the Display.
--- snip (z
Hi, All,
I run Asterisk with Digium hardware TDM 4 FXO & TE410P(E1).
When i dial PSTN to FXO or E1 PRI and softswich to SIP agent, the codec
always use
ulaw(TDM 4FXO) or alaw(TE410P).
Is it possible to change the default codec type to ILBC/GSM/G.729 when
PSTN -> SIP softswitch?
Thanks,
___
Thanks will do that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walt Reed
Sent: Friday, August 20, 2004 2:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] how to collect user entered digits
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Milli
Matt Schulte wrote:
Anyone have experience with Sipura's? Anyone know if they offer a
warranty? Would like opinions on these, good or flame.
We bought three SPA-2000s to use for our off premes extensions. Out of
those they all worked flawlessly until earlier this week when one just
stopped res
On Fri, 2004-08-20 at 12:52, Steve Szmidt wrote:
> Does ANYONE have music on hold working across IAX2? Google does not return
> anything on the subject. Except I did see on the release notes for 0.7.0
> "Better support for MOH in IAX2"
Yes. It works fine with no special config required.
--
> I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?
I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service codes are menti
> If you really want to be able to telnet in as root, locate
> telnetd.conf or somesuch and it should be in there somewhere
> as a yes/no. (It is for ssh anyway..)
No, not under any distro I'm familiar with... It's under /etc/securetty...
You add the tty of the device you want to allow root acces
Hi I try to install the h323 for asterisk but I get
this error, help please
make[1]: *** [asteriskaudio.o] Error 1make[1]:
Saliendo directorio `/root/asterisk-oh323-0.6.3a/wrapper'make: ***
[subdirs_all] Error 1
Best regards,
Diego Alberto Serrano BustosGerente
GeneralCall Me Networ
I am suprised that one would have to create a dialplan since its an already built in
function that works with regular POTS phones. Or is it because of the way DTMF is sent
via SIP?
> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
>
> What I envision is
Hello.
To answer your question, root is a restricted account. It
is too powerful to trust a telnet connection. So, you
telnet (preferably SSH) in as a normal user, and then type
`su -` and enter the root password. Su (short for
SuperUser?) allows you to become root. The - specifies to
load all
On Fri, 20 Aug 2004 13:52:57 -0400, Steve Szmidt wrote:
>-BEGIN PGP SIGNED MESSAGE-
>Hash: SHA1
>
>On Thursday 19 August 2004 08:35 pm, Robert Barnes wrote:
>> On Tue, 17 Aug 2004 10:19:17 -0400, Steve Szmidt <[EMAIL PROTECTED]> wrote:
>> > -BEGIN PGP SIGNED MESSAGE-
>> > Hash: SHA
> Wouldn't you need to track each extension? something like:
> exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
> exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> etc.?
>
Yep! good catch! that's why I asked someone to correct me, I was
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said:
> I have been searching thru all docs that I can find on wiki and such but can
> not get an answer. I am trying to collect a date from user input in the
> form of digits dialed from the phone to use in an agi script to do a
> database l
On Fri, Aug 20, 2004 at 10:13:16AM -0700, Chris Shaw said:
> - Original Message -
> From: "James Freire" <[EMAIL PROTECTED]>
>
> > I am using a Grandstream BT100 and I have been trying to get the PBX
> features to work for DND, call foward, etc. These functions do work when I
> use my POTS
On Fri, Aug 20, 2004 at 12:14:00PM -0500, Steven Critchfield said:
> On Fri, 2004-08-20 at 11:59, Steve Szmidt wrote:
> > > - Original Message -
> > > From: "Walt Reed" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Friday, August 20, 2004 9:13 AM
> > > Subject: Re: [Asterisk
On Fri, 2004-08-20 at 13:12, Brian McManus wrote:
> Tar ball it and make it accessible from a webserver of yours.
http://www.ekn.com/makecnf.tar
Comments welcome.
respectfully, Joseph ===
-= ** =
___
Asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 19 August 2004 08:35 pm, Robert Barnes wrote:
> On Tue, 17 Aug 2004 10:19:17 -0400, Steve Szmidt <[EMAIL PROTECTED]> wrote:
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> >
> > Hmm,
> >
> > My music on hold has always worked fine.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 20 August 2004 01:14 pm, Steven Critchfield wrote:
> On Fri, 2004-08-20 at 11:59, Steve Szmidt wrote:
> > > - Original Message -
> > > From: "Walt Reed" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Friday, August 20,
Hi Craig,
Thank you very much for the helpful information. I did enable that setting and it
seems to have worked but not all the way. I do a *72 for an unconditional call
forward + the number to forward to. Then when I dial the grandstream that has it
enabled, asterisk just reponds that the ex
Hello all,
I have been searching thru all docs that I can find on wiki and such but can
not get an answer. I am trying to collect a date from user input in the
form of digits dialed from the phone to use in an agi script to do a
database look up. I have tried to use "Get Data filename, timeout,
On Fri, 2004-08-20 at 11:59, Steve Szmidt wrote:
> > - Original Message -
> > From: "Walt Reed" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Friday, August 20, 2004 9:13 AM
> > Subject: Re: [Asterisk-Users] telnet and Root
> >
> > > On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chr
- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 9:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
> Hi All,
>
> I am using a Grandstream BT100 and I have been trying to get the PBX
features to work fo
Tar ball it and make it accessible from a webserver of yours. If you'd
like you can email me the tarball and i'll put it on an accessible URL.
Brian McManus
Joseph wrote:
On Fri, 2004-08-20 at 12:35, Ben Merrills wrote:
Sounds good, sounds like a handy thing to have around! :)
Ben
I don'
If you don't have somewhere to host it, drop me an email. Else yeah,
just stick it in the wiki, somewhere under the Cisco 79XX section?
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: 20 August 2004 17:45
To: [EMAIL PROTECTED]
Subject: RE
Hrm:
> [extensions]
> exten => 950X,Macro(siphandset,${EXTEN})
> exten => 951X,Macro(siphandset,${EXTEN})
> exten => 9521,Macro(siphandset,${EXTEN})
> exten => 9537,Macro(siphandset,${EXTEN})
> exten => 9543,Macro(siphandset,${EXTEN})
> ; ...
> ; define patterns matching all assigned extensions
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 20 August 2004 12:22 pm, Chris Shaw wrote:
> LOL it was so long ago, I didn't think about that reason... :)
>
> - Original Message -
> From: "Walt Reed" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, August 20, 2004 9:1
Ben Merrills wrote:
Sounds good, sounds like a handy thing to have around! :)
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: 20 August 2004 14:04
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Creating 79xx Configs
I made a little php scr
Can this be used with alaw audio files?
I have an AGI that generates alaw audio files, then
tries to use STREAM FILE to get asterisk to play them. The file is created in
/var/lib/asterisk/sounds and, if I put Background(filename) in the next
priority, it plays fine. I don’t quite know w
On Fri, 2004-08-20 at 12:35, Ben Merrills wrote:
> Sounds good, sounds like a handy thing to have around! :)
>
> Ben
I don't know where to post it?
I could not see a way to put it on the wiki...
--
respectfully, Joseph ===
-= ** =
__
> Is there any other way that we should be 'supposed' to do this
> functionality?
I would probably do something like this:
[macro-siphandset]
exten => s,1,Dial(SIP/${ARG1},16,tr)
exten => s,2,Voicemail(u${ARG1})
exten => s,102,Voicemail(b${ARG1})
; do whatever you want your normal SIP handsets to
See below
Jean-Yves Avenard wrote:
> Dear all.
>
> I've placed an order for several Uniden UIP200 SIP phone to connect to
> our Asterisk server but it seems that they're not going to be
> available for another while.
> The seller recommended the Ipdialog Siptone 2 instead which is a
> little bit d
Altus:
Go to www.pastebin.ca and copy up your config files and the error, then
paste the URLs to the mailing list. I'm sure we can diagnose the reason
or malconfiguration. (Don't forget to * out your passwords ;)
Brian
Altus Snyman wrote:
Good day all
I'm trying to configure 2 asterisk servers
Sounds good, sounds like a handy thing to have around! :)
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: 20 August 2004 14:04
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Creating 79xx Configs
I made a little php script that creates
Hi James,
This is a feature that needs to be enabled on both the phones and on
Asterisk. So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-
[9500]
context=internal
type=friend
username=9500
host=dynami
Hi, I was wondering how other people might handle this situation. We have a
10 channel fractional E1 and have so far only allocated about 25 numbers out
of a 100 number block (9500 - 9599). If an external caller dials an invalid
extension then we would like for them to be put through to our main
Matt Schulte wrote:
Anyone have experience with Sipura's? Anyone know if they offer a
warranty? Would like opinions on these, good or flame.
We bought *one* to test with and it died, can't even get a
response from Sipura "support". Could anyone recommend another device to
replace these? Prefer
LOL it was so long ago, I didn't think about that reason... :)
- Original Message -
From: "Walt Reed" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 9:13 AM
Subject: Re: [Asterisk-Users] telnet and Root
> On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chris Shaw sai
On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chris Shaw said:
> >...Today there's no valid reason to use telnet over ssh.
>
> Was there ever a valid reason? Maybe export restrictions on crypto? I've
> never EVER used telnet or rlogin, SSH is so much nicer anyway...
Yeah. Some of us were around befor
Hi All,
I am using a Grandstream BT100 and I have been trying to get the PBX features to work
for DND, call foward, etc. These functions do work when I use my POTS phones hooked up
to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP
phones. Is there a feature
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