Hi,
I would like to perform a Dial, and play an announce ment just once to
the caller. Actually very much like the A parameter, but this plays the
announcement to the callee instead of the caller.
I played around with MusicOnHold, for which I can specify a timeout, but
it doesn't continue in
I have this very specific problem with two dect sets. Problem that I
have is one-way audio, in this very rare situation.
I am calling with a Siemens N510 with C610 handset to Panasonic
KX-TGP500 with KX-TPA50 handset. This gives me problems when I am
calling to a SIP account that is configured
y idea's what to look for?
Thanks in advance.
Kind regards,
Roland.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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 *59.
But if
as well. Any other thoughts are welcome!
On Fri, Apr 13, 2012 at 10:36 AM, Roland wrote:
> Hi,
>
> What would be the easiest way to give a SIP account his own private queue,
> so calls can be added while this account is busy?
>
> While implementing our new Asterisk based telephone
interference 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
eems to be rather slow in processing the hint changes... so
it may take a few minutes before the changes take effect.
On Tue, Mar 27, 2012 at 1:25 PM, Roland wrote:
> I would like to fetch my extensions from the database. I created a dynamic
> hint, but doesn't seem to work. The BLF
so I guess I just have to be carefull, and
not mess around with callerid(num) too much.
On Tue, Mar 27, 2012 at 11:51 AM, A J Stiles
wrote:
> 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
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 e
using 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 g
e-
> From: 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 doe
ns 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-use
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 played
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,
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 wrote:
> thank you for your reply, but x-l
risk. Any other software
operator/switch boards 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 th
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---
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
this mailing list as well as the IRC channel that I'm not sure I could do
it again on my own..
so not to add more to my email, I'm seeking advice about the proper way to
learn about asterisk from A to Z if possible...
any advice would be appreciat
x27;s traffic QOS would surely 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-Commer
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
rin
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 side,
onversations overlapping
>
> RoLaNd RoLaNd wrote:
> >
> > Hi all,
> >
> > i'm facing this weird prob...my topology is as such:
> >
> >
> >
> > -
>
Hi all,
i'm facing this weird prob...my topology is as such:
-
-
when am on a call, sometimes when some1 else tries to call out.. i hear the
ac
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 A
> http://www.google.com/search?q=asterisk+authenticate&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a
>
> Thanks,
> Steve Totaro
>
> 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 password.. sorry if im not explaining
myself well.. heres an example:
i called my pstn
@306) ; gilberte
exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten => 303,1,VoicemailMain ; voicemail box to be redirected to
> Date: Thu, 21 Aug 2008 20:26:48 +0300
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digiu
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 th
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?!
___
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 a
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 authentica
@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
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
s
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
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:
sip
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 of
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 s
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 ste
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,
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 inc
sterisk-users@lists.digium.com
Date: Wed, 21 May 2008 10:29:21 -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 sur
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 pri
att 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/mak
> Message: 22> Date: Wed, 21 May 2008 06:49:39 -0700> From: Roberto Milani
> <[EMAIL PROTECTED]>> Subject: Re: [asterisk-users] asterisk and sipura 3102
> (pstn to> sip/sip to pstn calls)> To: Asterisk Users Mailing List -
> Non-Commercial Discussion
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 ou
Hi Carlos,
Check out Asterisk LDAP authentication:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
Greetz,
[EMAIL PROTECTED]
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Mohamed A.
Gombolaty
Gesendet: 27 February 2007 13:03
An: Asterisk Users Ma
...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
eivetext'...";
print "RECEIVE TEXT 3000\n";
my $result = ;
&checkresult($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-Co
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 2
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 softphones
Classical if-then functionality can be achieved in the dialpan with the
GotoIf(...) function.
Regards,
Roland.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von nik600
Gesendet: 23 December 2006 15:16
An: Asterisk Users Mailing List - Non
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
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
___
--Bandwidth and Colocation provided by E
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
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
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
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,
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
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
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 in
y PCI cards supporting 8 FXS interfaces, but
without success. does anyone know such hardware?
Thanks in advance,
Roland
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To UNSUBSCRI
, Peter, my dmesg command shows:
"TE410P version c01a009b"
Hope this helps you, 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
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
___
Asterisk-Users mailing list
Asterisk-
r a newbie to Asterisk.
Hope this helps!
Regards,
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 Asteri
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
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
from 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:
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,
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,
R
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,
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 a
he file
"asterisk"
and insert a "sleep 5" between stop and start in "restart".
hope this helps!
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Monday, July 04, 2005 9:06 PM
To: Aste
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 Discus
find it here:
http://www.digium.com/index.php?menu=product_detail&category=extras&prod
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 Discussi
i heard of people implementing this and it worked for them, i did not try it on
my own,
but you can only run with SCCP, not SIP! your seller might be right, i never
heard about
sip firmware for 7920!
i only have 7940 and 7960 phones running very fine with sip firmware.
regards,
roland
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
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
t;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] On Behalf Of Joseph
Sent: Monday, July 04, 2005 12:54 AM
To: Asteris
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 parameter
ck
command)?
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
toincomin
ed 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
Goo
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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http://lists.digium.com/mailman/listinfo/asterisk-users
To
not an asterisk issue, 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
Su
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
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
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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To UNSUBSCRIBE or update
ideas 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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System description:
Fedora Core 1
Kernel 2.4.22
Sudo 1.6.7p5
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
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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Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
Thanxxxx!
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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Asterisk
Hello! has anyone already successfully installed Digium TE410P card on
RedHat Enterprise Server 3.0?
Roland Zagler
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@fog smart partners
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#
U [peerIP]:5060 -> [myIP]:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
[myIP]:5060;branch=z9hG4bK069df2d9..To:
..From: ;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
[EMAIL PROTECTED]: [EMAIL PROTECTED]
90.238..Content-type: application/sdp..Max-Forwards:
70..Content-Length: 133.
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
=> _.,1,Answer
exten => _.,2,Dial,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
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
Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg41324.htm
l
Roland Zagler
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10,
Are you using kernel 2.6.x ?
Roland Zagler
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject: Re: RE: [Asterisk-Users] CAPI call transfer
Thanks for
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] O
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
Roland Zagler
mailto:[EMAIL PROTECTED]
mobile:4369910713694
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