Jordan Novak wrote:
Has anyone created a GUI for this.
I am not sure what youre looking for but we developed a Voicemail Manager:
=> http://sip-syndication.com
best regards,
Arnd
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asteri
Patrick Aljord wrote:
is there a good and free asterisk gui that is not tight to a live cd?
I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I
just want to run asterisk on my debian install. Is there a way to run
[EMAIL PROTECTED] on debian? or anything similar?
You can
Hi,
we are releasing an update of our "Tycho" Voicemail Manager. The update
to Beta 0.2 contains a bugfix and a couple of improvements over the 0.1 version:
Bug fix:
* missing "Channel Type" added to extension subscription
Improvements:
* adjustable refresh interval (voicemail)
*
Hi,
ive just installed a svn trunk (r42858) and i am having problems
getting app_voicemail to even try to connect to a imap server.
Ive added the following to voicemail.conf
--
; new IMAP Stuff
imapserver=mydom.com
imapport=143
expungeonhangup=no
[..]
[default]
; Office Accounts
7709810 => 1234,
Hi Benjamin,
Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not have http traffic on the same
box as asterisk.
Now, my qs:
Is th
Hi,
ive tried to setup a svn trunk version of asterisk to test
voicemail with imap support and i am so far without success.
Is there _anyone_ running voicemail with IMAP Support who can
answer some basic questions?
regards,
Arnd
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Hi,
Asterisk Mail List wrote:
I've integrated OpenVXI 3.4 (the latest one) with Asterisk for a
client. It is now in production, interpreting their VXML pages
using Asterisk for SIP/IAX telephony [..]
I also plan to release the code under the GPL as soon as I
can figure out the best way to do i
Hi,
is there anyone working on VXML or CCXML integration for asterisk?
If not, anyone interested in developing it?
-- Arnd
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
have a look at the nathelper examples in SER distribution. This is from
an rather old installation of mine.
--
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if lin
Andrea Spadaccini wrote:
are there any particolar guidelines to follow in order to make Asterisk
act as a SER client?
No. I have the following config:
register => account:[EMAIL PROTECTED]/asterisk-extension
and
[ser-out]
type=peer
secret=fump
host=serbox.com
callerid="MyMyselfAndi" <123456>
Siqhamo Sifo wrote:
I have ser sitting on my iptables nat box and my asterisk box on the lan .
Ser does forwarding so that any requests (register,invite,ack,...) to the
nat box at 5060 r sent to my asterisk box on the lan .I can register from
outside
to my asterisk box but there is only one wa
[EMAIL PROTECTED] wrote:
Is there a way to implement voicemail/email integration such that you
could retrieve the voicemail with either the phone or email, but only
have to delete the message once?
You can try our voicemail client called "Tycho", available for
MacOS X, Linux and Windooze. You
Arnd Vehling wrote:
Set this globally or in the virtual server config section. The scripts
wont work with most installations when safe_mode is off
^
wont work if safe_mode is ON!
Damn, not my day today.
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One more: be sure to set your webserver/php to register_globals off and
safe_mode off, example from apaches httpd.conf:
--
php_admin_flag register_globals off
php_admin_flag safe_mode off
--
Set this globally or in the virtual server config section. The scripts wont
work with most installation
Race Vanderdecken wrote:
Is all the messages or just a message now and then?
Sergio mailed me and said he cant listen to any voicemail. That was
a stupid bug in our php<>voicemail script and not related to the infamous
"orphant .txt" bug.
There was a bug fix in 1.2.10 for orphaned .txt files
Hello Sergio,
please download and install the vmxml scripts again, there was a problem when
php was configured with "register_globals=off". This is fixed now.
Please report success.
best regards,
Arnd
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Sergio R. D'Ippolito wrote:
I’m using Tycho software to see my voicemail, y can see de detail from
the message but i cant hear de message.
Please send me or post here:
- Client Version / os platform
- Server Operating System
- HTTP Server + php version
- which version of the scripts you downl
Steven wrote:
Good point.
If you click on that link twice by mistake, you may be deleting two different
voicemails.
As far as i tested it, its not possible because of the confirmation prompt
coming up and the to-be-deleted message will be removed from the selection
window before you can tr
Jordi Nelissen_ wrote:
did not yet took the time to evaluate Tycho, but the Voicemail XML
server side scripts are very helpfull. The source code is rather
self-explanatory ... but it could be even more helpfull if you could
provide us with some documentation on the usage of this API.
The doc
Hi,
were releasing a beta version of our java eclipse based
asterisk voicemail client called "Tycho". You can download the client
binary at: http://sip-syndication.com.
Please note that the client needs a set of php scripts installed
on the server side. The server side scripts, called "vmxml" ar
Hi There,
we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the "originate" command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the d
The "no sound" problem is very probably related to an incompatibility
between the zaptel driver and the Teles AVM B1 ISDN card i am using.
(kernel-capi 2.0)
Anyone knows a work-around for this or is there no way to make both
work?
cheers,
Arnd
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Just FYI for anyone else who might run into this problem:
After unloading the zaptel and zaprtc modules the audion works
again!
-- Arnd
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Hi,
i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07
and everything worked fine sofar when suddenly the voicemail and
musiconhold sound output stopped working.
The voicemailmenu still works though. I can see the voiceprompts etc
in the debug messages on the asterisk CLI but
Hi,
can anyone who has the Areski Calling Card solution on Asterisk
working comment on it? Is is stable enough for a production system?
Any pros and cons?
thx,
Arnd
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Hi,
apenon apenon wrote:
We have installed asterisk and using with many small ata. Now there is
an AS5300 outside a PSTN PBX which makes termination. Now I want
asterisk to handle only some of the calls with SIP and asterisk from
5300. There is no problem at asterisk configuration but in AS5300
Daniel Eboa wrote:
what i want is be able to authenticate user before they connected to my
asterisk box.
users can be registered with asterisk, but i want that each time a user want to
place outgoing call, he is first authenticate, and then authorize to place the
call through the asterisk box
Daniel Eboa wrote:
This is the scenario i want to setup:
Cisco ATA 186 ---> SER -> Asterisk
I want the Cisco ATA to register to Asterisk through SER. when the Cisco
ATA place a call, SER querry a data base (MySQL or else), and if there
is a valid Account for the ATA, the
Dan Morin wrote:
Yes, I have both nat=yes and canreinvite=no.
I have similiar setting (nat=route, canreinvite=no) and ive seen the same
problems. My Server is on the internet though. I dont use any NAT support
on the GS side and it does work most of the time. I havent seen this issue
with my
Hi,
Terry H. Gilsenan wrote:
I was having this problem with Gradstream BT101's with Asterisk @ Home
version 0.7.
The problem was that there was a sip channel still open (as far as asterisk
and the phone were concerned) however this sip channel was not actually in
use. The existence of this sip
C F wrote:
are these phones behind nat?
Yes, but correctly registered. The same fones dont have any problems
when registered to a SER Server.
Can constantly reloading the configuration cause problems?
cheers,
Arnd
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Hi,
some of my sip fones which have several external numbers assigned
are not reachable after a certain timespan. Instead of the fone the
Voicemailbox is trigger in "busy" mode. After a reboot if the sip-fone
the problem goes away for some time. Ive seen this problem with Sipuras
and Grandstreams
Hi,
thats very probably a NAT problem. Your NAT box is probaly blocking
the incoming UDP voice stream.
If asteriks supports a RTP Proxy you can try that.
best regards,
Arnd
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Hi,
is it possible to use Asteriks for translating SIP to H323 and vice versa?
I am looking to implement the following Setup
SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC
Basicly i want SIP fones to talk to H323 fones and and SIP Fones to
access PSTN Gateway(s) in a H323
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