anyone seen or used * voicemail on a PBX system that needs to talk SMDI
to the VM host?
Dave Packham
Dave Packham
University of Utah Netcom
Campus RD
c. [EMAIL PROTECTED]
w. [EMAIL PROTECTED]
[EMAIL PROTECTED]
Trillian/ICQ#:45818442
MSN [EMAIL PROTECTED]
Our Groups Website
http
VOIP services are illegal I thought in the Bahamas according batelco
:)
What island are you on?
Have fun finding some.
Dave P
[EMAIL PROTECTED] 2/24/2004 4:16:59 PM
Need SIP or H.323 origination from Bahamas ASAP. Can someone
provide an
access number or Toll Free number origination
can we get a copy of your saved configs?
Dave P
[EMAIL PROTECTED] 3/15/2004 10:12:10 AM
I gave my test * server away, so I can no longer test it, but I do
have
copies of
my configs. I did not make extensive changes to get it to work. I
was
using a
slightly older oh323 release (0.5.6), but
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
and phone.conf with this email.
Thanks again
Dave
[EMAIL PROTECTED] 3/22/2003 9:35:45 AM
On Friday 21 March 2003 22:55, Dave Packham wrote:
I have a linejack and a phone jack in my asterisk server
working well between the SIP phones and the phonejack. what I
cannot get to work is the outbound
ok I changed that but I still get a busy after the 6 digit. if dialing
95551212 I get a busy on the last 1 and not after the last 2
Thanks again :)
Dave
[EMAIL PROTECTED] 3/24/2003 12:45:35 PM
...
I still get a busy when I hit the 6th number of
a
7 digit dial +9 for the outside line?
has anyone ever used an 827-4v DSL router to do SIPtoPOTS conversion? how would I set
up my 4 pots lines to be SIP extensions/phones? any ideas? Cisco site is a little
lacking on sample configs. I would like to set up 3 of the ports for FXS analog sets
and one port for FXO. convert all the
does anyone have a recorder GSM file that emulates the Telco's if you are a
telemarketer please hangup now recording? I don't see one in the sounds dir. the
ZapATEller works great for computerized callers but if a human hears this message
asking them to go away they have to. Isn't that
(sort of) that qwest charges $8 a month to
play and makes it illegal for the telemarketers to continue as they have recieved
notification that you want to be removed etc.
Still looking :)
Dave
Dave Packham
University of Utah NetCom
Manager R D
D.S.O.
University Incident Response Team
c
who in the US sells these?
I cant find anyone listed in google.com.
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Cisco has just sent me a email that said the Cisco 7905 will have SIP
running soon. the image is out internally and will be on the Cisco site
soon.
I got this from a TAC person. dont know if its the truth until I can
DL the image... :)
Dave
[EMAIL PROTECTED] 6/25/2003 2:03:46 PM
The Cisco
to release the code to Mark soon. Any comments welcome, just
look for me on IRC (p0lar) or use the list.
Here is the link
http://warpcore.netcom.utah.edu/openconf/openconf.php
Dave Packham
University of Utah NetCom
Manager R D
D.S.O.
University Incident Response Team
c. [EMAIL PROTECTED]
w
Check to see if you can get a IOS code leverl that supports SIP on the
6500. then maybe you can use your E1 card directly. you can also get a
SIP version of the code for the 7960's etc
Dave
[EMAIL PROTECTED] 6/28/2003 2:56:12 PM
Hi Chris
I've done a lot of things with Cisco AVVID solutions
do your format changes allow support for raw wav files to played as
prompts?
Dave
[EMAIL PROTECTED] 6/28/2003 6:05:20 PM
Uh, what are we looking for other than better playback performance?
That I didn't break anything. The real cool stuff should be available
fairly soon when I finish the
Does anyone know if someone makes a hard video phone for SIP.
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Same prob here. 15 SIP phones only get eco when going to the PSTN...
if you find something let me know
Dave
[EMAIL PROTECTED] 7/1/2003 8:53:13 AM
Hello,
I can't have asterisk working without echo when I place a call from IP
phone (SIP or H323) to a PSTN Phone. The called number as no
The * code is not written yet. The Digium's cards rock... (ps I also
have a linejack in my drawer)
Dave
[EMAIL PROTECTED] 7/3/2003 2:10:23 AM
What do you mean a feature that is not present? I can dial out with
other apps...
-Z
- Original Message -
From: Andres Tello Abrego [EMAIL
So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I
cannot get the phones to talk/RTP to each other. jtodd has had this problem in the
past with the 186's. Just wondering if anyone has a reason why Cisco sometimes poop
on reinvite is the Cisco code broke? if so
anyone using a SIP based video phone with * yet? I would like to buy some but would
like it to work with * first
Thanks
Dave Packham
ie p0lar
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I would like to see your code...
sounds great
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I am having the same probs. I get local dialing tones but no audio after the call is
connected.. I got a private build from Xten and it was the same
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Fixed it I have audio now... uninstall everything xten makes and manually clear
out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the
keys. reinstall Xpro and it works... go figure
Dave
___
Asterisk-Users
All
I have given the * PHP web interface files to Mark to check out. hopefully he will
include them into the CVS tree soon.
Dave Packham
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
is there any way to keep those vars around until after h goes away?maybe move the
free routiene to after h is done?
Dave
[EMAIL PROTECTED] 7/25/2003 5:32:55 PM
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with
just be a message thing on * server.
Dave Packham
[EMAIL PROTECTED] 7/28/2003 4:16:16 PM
On your sip.conf for each sip endopoint set canreinvite = yes.
That way the rtp stream won t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT
can you do the stutter tone on Multiple SIP voicemail extensions? or only one
extension listed in the zapata.conf?
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
special though:
[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband
[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband
-Original Message-
From: Dave Packham [mailto:[EMAIL
-
From: Dave Packham [mailto:[EMAIL PROTECTED]
Sent: 29 July 2003 15:43
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk
server ...
can you share the SIP conf entries that you are using to get
this to work? I have played
I had M$ Mess working a bit ago but now I cant seem to make it work. can someone post
whatI need in SIP.conf for a comfig to get it working again?
Thanks
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
ok now lets modify that mix script to pick up on who started the monitored call and
look them up in the voicemail.conf and email it to em
Dave
[EMAIL PROTECTED] 8/25/2003 2:14:16 PM
Note that h will not be called if you park the call and pick it backup.
bkw
On Mon, 25 Aug 2003, David Harris
Audio Sample Rate 8 kHz
Audio Format PCM
I dont really think that the monitor files are getting GSM'd correctly.
Ill RTFM on sox and see what I can find
Dave
[EMAIL PROTECTED] 8/25/2003 3:41:07 PM
My mux script does the gsm compression using sox
On Mon, 25 Aug 2003, Dave Packham wrote
Thanks for the reply. snips of it are in the Cisco TAC case logs and developers are
looking at it. Ill let you know if I get a resoloution
Dave P
[EMAIL PROTECTED] 8/23/2003 12:53:11 PM
Normally the caller-id is taken from remote-party-id in the SIP
INVITE. We don't see that field
http://www.nero.com/us/631911127302064.html
Have you all seen this?
Its a SIP softphone put out by the people that do the CD burning software Nero...
Check it out it works with *
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
several SIP webpages list the 2004i as a SIP hardphone. I have 50+ of them that I
would love to use with * but cant one Nortel rep said they are writing a flash for
it and one said not. we can only hope
Dave
[EMAIL PROTECTED] 9/9/2003 10:09:23 PM
Where did you hear that they
nope
when I click on something on the left I get a FQDN not just the pne you had
Hmmm.
can you give me more info or can I look at your site directly? from the outside?
Dave
[EMAIL PROTECTED] 9/11/2003 8:55:27 PM
On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote:
hmm
Ill be writing a README and INSTALL tonight and getting that into CVS to
http://rads.netcom.utah.edu/phpconfig/phpconfig.php
Lemme know if you have any patches or add on's are welcome
Dave Packham
aka
p0lar
___
Asterisk-Users mailing list
I have put my phpconfig stuff out into the Digium CVS tree.
Project name is
phpconfig.
see it at
http://rads.netcom.utah.edu/phpconfig/phpconfig.php
Lemme know if you have any patches or add on's are welcome
Dave Packham
aka
p0lar
at 10:12:50PM -0600, Dave Packham wrote:
nope
when I click on something on the left I get a FQDN not just the pne you
had
Hmmm.
Further info: it works with Microsoft Internet Explorer. It
does not work with Mozilla 1.4 under Linux. It also does
work with Mozilla Firebird
wrote:
On Thu, Sep 11, 2003 at 07:57:58PM -0600, Dave Packham wrote:
I have put my phpconfig stuff out into the Digium CVS tree.
Project name is
phpconfig.
see it at
http://rads.netcom.utah.edu/phpconfig/phpconfig.php
Looks cool, but the links don't work
in the phpconfig_init.php you need to make sure that the files paths are correct
p0lar
[EMAIL PROTECTED] 9/28/2003 1:17:28 PM
Hi,
Just giving phpconfig a try but can't find and setup instructions..
What I have done so far..
1. Copied the phpconfig files to the web dir on the server.
2.
I have commit access to the phpconfig cvs on digiums site... Mark and team have been
great to work with. I would be happy to work on changes to the phpconfig to add cdr
work or anything. its just a framework that needs wizzards, reports etc to be
finished
Dave
p0lar
[EMAIL
. jump from config to config etc its a framework to get
people thinking about abbing a web interface to *.
Dave
aka
p0lar
Dave Packham
University of Utah NetCom
Manager RD
D.S.O.
University Incident Response Team
c. [EMAIL PROTECTED]
w. [EMAIL PROTECTED]
FWD. 20223
[EMAIL
We are looking into using the Linux SMDI code to write a * app module.
have not gotten far.. but we could help... with the Linux SMDI stuff
already written it shouldn't be too hard.
http://rpmfind.net/linux/RPM/contrib/libc6/i386/smdi-0.0.3-1.i386.html
like that
Dave P
[EMAIL PROTECTED]
has anyone done caller announce in MeetMe's before?
Dave P
[EMAIL PROTECTED] 5/18/2004 5:50:49 PM
With multiple parking lots you can give each person their own lot thus
exten
800 for everyone will connect them with just their call passing the lot
name
which you know for X customer.
bkw
-
have you tried the #asterisk-dev IRC room? thats the best place
Dave P
[EMAIL PROTECTED] 5/19/2004 2:12:10 AM
In article [EMAIL PROTECTED],
Dave Packham [EMAIL PROTECTED] wrote:
has anyone done caller announce in MeetMe's before?
I'm working on some modifications that should make
I can provide logins and dev env to an asterisk server with an SMDI
serial connection to anyone willing to work on the SMDI bounty.we
have looked into this and got the hardware setup but I dont have time to
write the code..
Dave P
[EMAIL PROTECTED] 5/25/2004 1:45:12 PM
W. Kevin Hunt wrote:
Just a php config file interface. check out phpconfig in cvs. its just for editing
and
parsing the conf files
Dave
[EMAIL PROTECTED] 11/26/2003 7:33:04 AM
Does anyone know if a web interface has been created for * ?
--
*
Not everyone is touched by an Angel
Those that are,
what would be nice is to get this on MeetMe app. so that you can announce someone
joining the conf call
Dave
[EMAIL PROTECTED] 12/1/2003 11:11:49 AM
--- Vledder, Hans [EMAIL PROTECTED] wrote:
I would like to play an announcement to the user on what external line a call came
in, right before
what do the options algo do in the monitor app? I dont see that in the show
application monitor? is this a patch?
Dave
[EMAIL PROTECTED] 12/2/2003 6:56:18 AM
Try something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps
I have 40 of these phones. they dont run SIP or any usable protocol they can
hook up to a Nortel box and proxy SIP out of that box, but they wont run SIP
native if im wrong please let me know... I'd relly like to use my 40 phones that
are collecting dust
Dave
[EMAIL PROTECTED]
I have a 4 port card in a regular system and I get that prob sometimes when I copy
large files to that server IRQ problem? if I stop the copy the sound prob goes
away. not a big help but at least you know that your not alone...
Dave..
[EMAIL PROTECTED] 12/15/2003 10:17:35 AM
On Mon,
is it possiable to get a var to show user name and number in the sendmail from line?
I see a var for the callid num but not name. is this something I need to write?
Dave P
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
52 matches
Mail list logo
