I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how to
make it work by myself.
any other very useful new guides you guys have? tnx
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that really work. Maybe there's only a few that
really stand out?
I have also checked Voice Operator Panel on the website. Any reviews on
that?
I hope somebody would like to share their setup.
Thanks in advance!
Kind regards,
Roland
I managed to do that once by using another SIP account, for example at
Voipbuster. It's free. Once you are connected, you can still use ext@IP of
your server. I guess you could use any other free SIP account.
On Tue, Jan 3, 2012 at 4:01 PM, Faraj Khasib fkha...@iconnecths.com wrote:
thank you
iptables -L -n | grep icmp gives you the same on both machines?
Is it possible that the other public IP is behind a main firewall,
provided by your ISP? I know our hosting provider has this. They filter all
traffic through their main router, and after that locally with iptables.
On Tue, Jan 3,
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be
will generate their own Answer() if not present, others will not.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Roland
*Sent:* Monday, January 16, 2012 9:22 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
Ok, got it. Indeed
something like:
callerid=137-Roland 31229253137
137 would be my extention number here.
I think the downside of this is, that I should configure this for each SIP
account. I could specify a default callerid, which our main DID, in a
template, but then people will see this general ID when I call internal
I would like to fetch my extensions from the database. I created a dynamic
hint, but doesn't seem to work. The BLF on my phone doesn't change when the
state of the extension changed.
This is in my dialplan:
exten = _ZXX!,hint,${SIP_BYEXT(${EXTEN},${CONTEXT})}
exten = _ZXX!,1,Verbose(3, Search
with callerid(num) too much.
On Tue, Mar 27, 2012 at 11:51 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Tuesday 27 March 2012, Roland wrote:
I am setting up my dialplan with quite some outbound numbers. We have a
block of 100 DID's, for which some of them will go direct to specific
before the changes take effect.
On Tue, Mar 27, 2012 at 1:25 PM, Roland aster...@rolandow.com wrote:
I would like to fetch my extensions from the database. I created a dynamic
hint, but doesn't seem to work. The BLF on my phone doesn't change when the
state of the extension changed
on
the signal.
So I think I should solve this with Asterisk. Any suggestions about
queueing call to a extension (or SIP account actually) without having to
configure a 'private' queue for each sip account?
Thanks in advance!
Kind regards,
Roland
We have a extension pad on our Yealink phone for the receptionist. With our
old non-voip PBX system, the receptionist could pickup a specific extension
by pressing the corresponding key. Is this possible with Asterisk too?
I have configured Asterisk to pickup a specific extension with *59exten.
to look for?
Thanks in advance.
Kind regards,
Roland.
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Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call
: Re: [asterisk-users] asterisk and sipura 3102
(pstn to sip/sip to pstn calls) To: Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID:
[EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252 Hi
Roland I have 2 linksys spa-3102
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNdSent:
Wednesday, May 21, 2008 9:01 AMTo: [EMAIL PROTECTED]: [asterisk-users] asterisk
and sipura 3102 (pstn to sip/sip to pstn calls) Hello all, its been a while im
trying to setup my asterisk/sipura 3102 to recieve/make
Hi Jose,
i just did that, doesnt seem to work..
its still giving me the same error
Date: Wed, 21 May 2008 11:02:36 -0500
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to
pstn calls)
I was seeing your
-0700
Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to
pstn calls)
Ciao Roland
your dialplan:Exten = _1XX,1,Dial(SIP/${EXTEN})
_1XX is a three (3) digit number starting with 1, I'm not sure what happens if
you dial 1009 but it seems that it is dialing.
Anyway
Hello all,
ive got the following setup currently:
__Sipura 3102-PSTN
|
Lan |
|
|__asterisk
i configured both asterisk and pstn to be able to receive/make calls through
each other using sip of course..
but the thing is i want asterisk that when it receives an
Hey thanks for the help :)
though i already did that, and the sip debugging info shows me tht its ringing
on the respective sip extension (1002) but nothing is happening..
so i guess its true it IS a diala plan issue tht i am yet to figure it out ...
Date: Sat, 24 May 2008 14:20:45 +0100
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.i want
to add a
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.i want
to add a
hello all,
was wondering if some1 could help me to add an option to my incoming operator
menu.
currently, when some1 calls in, he gets a recorded msg asking for him to punch
in an extension or dial 100 for operator assistance wht i want is to add 2
other things;
firstly, if in a period
Hi all,
i've recently acquired a callcentric account.
i've perfectly setup my sip.conf and extensions.conf to make outgoing calls.
but the problem is with incoming calls! when i call in, asterisk doesnt even
see the incoming call!
how is tht possible!
please see the following my config:
Hello all,
i recently finished setting up my asterisk with sipura 3102 using PSTN.
this is my dial plan relevant to wht i want:
exten =_01,1,Dial(SIP/$(EXTEN)@200)
right now as u see i made my dial plan on a 2 stage dialing mode.
tht means i dial 01, i get the pstn dial tone, and then i call
Hey!
i'm facing the same prob..
i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip
client..!
so far i found these 3:
AGEphone mobile: http://www.ageet.com/
SJphone: http://www.sjlabs.com/sjp.html
Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html
@lists.digium.com
Date: Thu, 3 Jul 2008 13:11:40 -0400
Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?
Hi Roland,
Did you try:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk
hi all,
is there any way of removing this line from showing on the console?
my verbosity level is 3.
and this is the following output on cli 24/7 unless its interrupted by the msgs
tht really counts like connected sip and so on..
[Jul 4 10:32:38] NOTICE[18542]: manager.c:1015
Hi All,
i have asterisk with 9 SIP accounts on it.
i was wondering if theres a way to setup asterisk, to send the amount of
minutes each SIP account have spent incoming as well as outgoing and if
possible the number it called!
any advice?!
any help would truly be appreciated..!
thanks in
Hello all,
i read a few articles online about the possibility to setup a buzzer door
system to PBX using asterisk!
currently my setup contains asterisk of course, and a sipura 3102..
what do i need to get such a feature done?!
or should i ask if its possible?!
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with softphones
all over the house!
could someone help me set up a limitation for my wife and kids not to be able
to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way to
deny a specific extension (300) from calling another specific extensions (03)
except if the caller punch a specified password.. sorry if im not explaining
myself well.. heres an example:
i called my pstn
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way
to deny a specific extension (300) from calling another specific extensions
(03) except if the caller punch a specified
i kinda have a relevant prob!
my sipura 3102 wont pass CID to asterisk even though ive enabled such a feature
in its web gui!
Date: Wed, 27 Aug 2008 12:07:51 -0600
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Off-Hook (type II) CID passing to
Hi all,
i'm facing this weird prob...my topology is as such:
softphone --- asterisk sipura 3102
sipura 3102
-sipura 3102
-sipura 3102
when am on
RoLaNd RoLaNd wrote:
Hi all,
i'm facing this weird prob...my topology is as such:
softphone --- asterisk sipura 3102
sipura 3102
-sipura 3102
i appologize for not making myself clear..
i have my asterisk box, connexted to 4 sipura3102..
these sipuras has 4 PSTN lines connected to them through one cable, which has
8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve
fxs port in the sipura)
on the other
first of all my topology is as such:Softphones-- asterisk --
sipurasoftphone with peer number 100, calls another softphone with peer number
as 200. (both has asterisk as gateway)relevant extensions.conf:
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3
help with no doubt.. this
would decrease latency as well
hope I've shed some light about this, if not well the more knowledge the betteR
best,
Roland
From: michel freiha
Sent: Monday, December 15, 2008 10:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users
Hi all,
a few month ago I got the task of setting up asterisk for my company.
I had 94 employee to set this up for ...
I never heard of asterisk before to b honest, so after researching a bit..
I started with a digium card with ZAP
though that didn't work out as the card were flawed..
so ended up
Hello,
i've recently configured my asterisk for internal sip calls.
while testing, i noticed that 1 out of 10 calls works..
at first i thought my router dropping packets around the way as it were a
bottle neck..
so i've added a switch.
once i tested again same prob occurs...
im using xlite as
or workaround for this problem?
Best regards,
Lars Roland
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, the optipoint is not registering correct to asterisk, when it is
connected to the network.(By the way, SJphone does register to asterisk)
So here is my question, does anyone suffer from the same problem and/or solved it???
Thanks a lot,
greetz
Roland (erlangen/germany
:-(
Do you use another Hardphone ?
Thanks in advance !
Roland
Hi!
I have updated the optipoint to the last software version
I can Call the optipoint from other phones and talk.
The optipoint register with asterisk but in the phone display i have
only no server. and no dial
, but I don´t know how long.
Thanks for your help !
regards
Roland
Hi!
Yes we have many kinds of phones hwere in the show room, snom, polycom,
cisco, grandstream, ipdialog, intracom, d-link, symbol all of them works
with asterisk with some testing and with some issues ...but works
],30,r)
Put something in front of the ${EXTEN:0} ???
For incoming calls, I already did a few lines that the user roland.knoerl
is available by dialing 772.
Thanks in advance !
Roland Knörl , Nuremberg , Germany
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[EMAIL
!
greetings
Roland, nuermberg, germany
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],30,r)
Put something in front of the ${EXTEN:0} ???
For incoming calls, I already did a few lines that makes the user roland.knoerl
available by dialing 772.
Thanks in advance !
Roland Knoerl , Nuremberg , Germany
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how i could fix this without
changing the OS! I could use a newer Kernel but then it runs out of
support at RedHat.
Thank you in advance,
Roland
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Hello!
Can i only use one gatekeeper in OH323? Is there any documentation about
how to use gatekeeper-ids?
Thanks,
Roland
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Hello list,
does anyone know how to change the interdigit timeout when using Cisco
IP Phone 7940/7960 with SIP-Firmware and Asterisk?
it's default value is 15 sec. but i have nothing found to set this in
tftp-config file etc.
Thanks in advance,
Roland
try the cisco 7940 with sip firmware:
tons of features and easy to install
see http://www.voip-info.org/tiki-index.php?page=cisco%2079xx
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexandre
Leclerc
Sent: Tuesday, July 12, 2005 6:16
Hello list,
is there anyone out there that could grab the new SIP firmware 7.5
for the 7940/7960 from Cisco's Site and mail it to me ([EMAIL PROTECTED])?
i already ordered a support contract but did not get my access data yet!
Thanks,
Roland
Thanks,
i added
dialplan_template: dialplan
to SIPDefault.cnf and the lines you sent to
dialplan.xml in TFTP-directory and it works!
Thanks again,
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Wellsted
Sent: Tuesday, July 12, 2005 7:08
cisco
that are VoIP capable.
this and everything else can be found by experiencing the search
button on the wiki-site...
best regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Pastore
Sent: Tuesday, July 12, 2005 11:33 PM
To: asterisk-users
Hi Freddy,
we use the drivers from RedHat Enterprise Linux 4 and they work great.
i think it depends just on the kernel version.
e.g.
http://h18000.www1.hp.com/support/files/server/us/locate/1116_6011.html
for the DL360
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto
is /usr/local/sbin/mailfax flagged to 755?
Von: [EMAIL PROTECTED] im Auftrag von Rob Danz
Gesendet: Mi 13.07.2005 17:17
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] SpanDSP rxfax, no tiff.
Hello,
Let me start by saying I have checked the wiki
,
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Felder
Sent: Thursday, July 14, 2005 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Cisco 7960 on Asterisk?
Hello,
I am just building my first
Hello list,
Did anyone already get the T410P card running in an
HP-Compaq DL380 G4 server? If yes, how?
I'm using Fedora Core 3 with 2.6.11-1.35_FC3smp Kernel package.
Thanks in advance,
Roland
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, folks!
Best regards,
Roland
-Original Message-
Sent: Wednesday, July 20, 2005 10:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: Roland Zagler
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
server
What is your dmesg output when you fire up
cards supporting 8 FXS interfaces, but
without success. does anyone know such hardware?
Thanks in advance,
Roland
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Thanks for the hint, where have you bought them?
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Tuesday, August 16, 2005 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 8 FXS
Hello,
i was wondering if it is possible to execute an AGI or shell script when
a channel is answered. Does anyone have suggestions on how to do this?
Thanks in advance,
Roland
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http
Thanks for the hint, do you know where to buy it (cheap) and the
price for it?
Thanks,
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP
Newbie
Sent: Wednesday, August 17, 2005 6:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Thanks,
Roland
Roland Welker
Moray Office Supplies
Edgar Road, Elgin, IV30 6YQ
T: +44/(0)1343/549869
On Thu, 2005-06-23 at 21:43 +0800, Steve Underwood wrote:
Roland Welker wrote:
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Unless
, you should take a look at the manual of
your
email system on how to create groups.
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Burd
Sent: Saturday, July 02, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users
be connected to SIP Phone
100
any suggestions on how to implement this in an easy way?
Thanks in advance,
Roland Zagler
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and the announcement has been
played. Before
connecting to SIP Phone 100 the caller should hear a soundfile...
wiki says nothing about an Dial-option to play a soundfile to the caller
;-(
Roland Zagler
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent
)?
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Saturday, July 02, 2005 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] play message to callee before connect
toincomingcall
yes, robert, but how do i join the two legs inside a call file or
in the dialplan?
i have experienced that call files can do a call to a channel and
if this call is answered it can either be connected to an extension
inside a context or call an application with parameters.
roland
-Original
)
extensions.conf of server2:
exten = _1X.,1,Dial(IAX2/server1/${EXTEN:1},30)
use deny and permit only with later versions than 1.0.5 of asterisk
(best with CVS HEAD)
i hope this helps
best regards,
Roland Zagler
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Thanks for the suggestion, C F, but the problem is there is a rather big
database application behind with many users, so a static configuration
is not suitable for my needs. i am working mostly with realtime and agi.
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto
Sure!
http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2
regards, roland
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoglu
Sent: Monday, July 04, 2005 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
find it here:
http://www.digium.com/index.php?menu=product_detailcategory=extrasprod
uct=G729
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis
curty
Sent: Monday, July 04, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial
did you use the zaptel drivers? you need a timer interface for meetme
application! use ztdummy!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed
Farid
Sent: Monday, July 04, 2005 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial
start
during bootup process all is started fine.
i experienced that on some configs the service asterisk restart does
not work correctly, so go to /etc/rc.d/init.d and edit the file
asterisk
and insert a sleep 5 between stop and start in restart.
hope this helps!
regards,
roland
-Original
to not registering to asterisk an
incoming call could not be delivered.
HELP PLZ. ! ;-)
Kind regards thx for help in advance
Roland / Nuermberg / Germany
P.S. Wendys können wir mal mails auf deutsch austauschen. wenn du schon aus nürnberg
bist :-)
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Has anyone experienced in connecting a asterisk pbx to douglas telecom
successfully? If yes, could you please post your SIP.CONF and your
EXTENSIONS.CONF!
Thanx in advance,
Roland
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http
Try specifying your number you want to dial with b in front of, e.g.
Dial(CAPI/01824708169:b01824708752,60) in your extensions.conf!
Regards,
roland
Roland Zagler
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent
Can you post your extensions.conf, maybe i can find something!
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM
To: [EMAIL PROTECTED]
Subject: Re
,CAPI/50:b${EXTEN},60
exten = _.,100,Hangup
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:03 PM
To: [EMAIL PROTECTED]
Subject: Re: RE: RE: RE: RE
You could try to specify incomingmsn *NOT* to * and outgoingmsn in
your capi.conf
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:38 PM
To: [EMAIL
..Call-ID:
[EMAIL PROTECTED]: [EMAIL PROTECTED]
90.238..Content-type: application/sdp..Max-Forwards:
70..Content-Length: 133v=0..o=none 0 0 IN IP4 [peerIP]..s=-..c=IN
IP4 198.31.231.1
7..t=0 0..m=audio 18691 RTP/AVP 8..a=rtpmap:8 PCMA/8000..a=ptime:30..
Than
Roland Zagler
mailto
Hello! has anyone already successfully installed Digium TE410P card on
RedHat Enterprise Server 3.0?
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Hello!
Is it possible to run Asterisk as a SMS Service Center and does it work
over a directly connected ISDN (CAPI) interface card?
Did anyone already use Asterisk for that?
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
Than!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Hello!
Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?
Thanx in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Apache httpd 2.0.50
Asterisk 1.0-RC2
Can anyone please help?
Thank you in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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such cards?
Specifications:
PCI or MiniPCI
up to 120 concurrent transcodings
Codecs: G.729/G.729A or G.723.1 or GSM or combinations of them
Thank you in advance,
Roland Zagler
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Asterisk-Users
I'm not familiar with Quintum, but what codec do you mean at the allow= line
in sip.conf
with h723?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Freddy Setiawan
Sent: Sunday, June 25, 2006 8:37 PM
To: asterisk-users@lists.digium.com
Subject:
extensions may be managed with a separate
account type Asterisk extensions.
It would be great if some of you could test this and write me your
feedback via email.
- --
Best regards
Roland Gruber
LDAP Account Manager
http://www.ldap-account-manager.org/
-BEGIN PGP SIGNATURE-
Version: GnuPG
Try latest IAX2 YakaPhone which you can get from www.yakasoftware.com.
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von ismail loo
Gesendet: 05 February 2007 17:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] IAX2
Hi Mark,
Take a look at the YakaVOIP solution from http://www.yakasoftware.com/
http://www.yakasoftware.com. Probably suits your requirements.
Greetz,
Roland.
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von MBIT
Technologies
Gesendet: 12 February 2007 22:23
;
checkresult($result);
Greetz,
Roland.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Olle E
Johansson
Gesendet: 24 February 2007 10:52
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] ReceiveText()?
24 feb 2007
...You can declare a variable whose values gets set/used anywhere in the
dialplan.
Regards,
Roland.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Yuan LIU
Gesendet: 25 February 2007 08:41
An: asterisk-users@lists.digium.com
Betreff: RE: AW
Hi Carlos,
Check out Asterisk LDAP authentication:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
Greetz,
[EMAIL PROTECTED]
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Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Mohamed A.
Gombolaty
Gesendet: 27 February 2007 13:03
An: Asterisk Users
Hi,
Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the secret AND/OR md5secret columns always have to
contain the password in plain text even when you set the auth column value
to md5?!?
Am I missing out something? Any ideas on how to correct this?
Hi,
Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the secret AND/OR md5secret columns always have to
contain the password in plain text even when you set the auth column value
to md5?!?
Am I missing out something? Any ideas on how to correct this?
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