Displaying 20 results from an estimated 5000 matches similar to: "Announcement to caller when called party has picked up - without initial Answer()?"
2005 Jan 31
3
Announcement to caller when called party haspicked up - without initial Answer()?
> -----Original Message-----
> From: David Liu [mailto:david@deltapath.com]
> Sent: 31 January 2005 14:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Announcement to caller when
> called party haspicked up - without initial Answer()?
>
>
> This is super easy to do. All you need to do is to put that
> announcement
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2006 Sep 14
3
One way audio problem on gateway to PSTN after some time, no NAT involved
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN
What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called
2004 Jul 12
1
R: How to make * don't strip the leading 0
> Is it possible to tell asterisk not to strip the leading 0
> of *incoming* MSNs? I use asterisk with i4l and whenever
> I get a call from an long-distance party, the leading 0, which
> should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest
2003 May 06
2
active ftp & connection tracking ?
this :
iptables -A FORWARD -i internal-interface -j ACCEPT
iptables -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT
iptables -A FORWARD -j DROP
doesn''t seem to work for active-ftp .. i even manualy loaded ip_conntrack_ftp but as u see it is unused :
# lsmod
Module Size Used by Not tainted
ip_conntrack_ftp 4272 0 (unused)
iptable_nat
2002 Jul 28
1
"For ethernet, no packet uses less than 64 bytes" - why?
Hi
Well, subject says all. In Chapter 9.2.2.1, TBF, the parameter mpu
or "minimum packet size" is explained as:
> A zero-sized packet does not use zero bandwidth. For ethernet, no packet
> uses less than 64 bytes. The Minimum Packet Unit determines the minimal
> token usage for a packet.
In my understanding an ethernet packet needs at least 14 (2*6+2) bytes or
54 bytes if
2004 Jul 12
3
How to make * don't strip the leading 0
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile phone
0177-1234567 should be displayed, but 177-1234567 is displayed. I double
checked if I've forgotten to remove an
2005 Jun 29
2
Play an announcement to the CALLING party
Hi folks,
how could I play an announcement to the calling party as soon, as the
called party picked up. I would like to deploy an asterisk in an
environment, where a premium rate support-number is offered to customers
which do not want to pay a monthly support contract. In Germany, you are
commited by law to announce the cost per minute of a premium rate number at
the beginning of the call. So,
2014 Apr 11
1
SIP fraud IP blacklist
Hi,
in case, anyone is interested...
I have started compiling a blacklist of hosts and networks from which
SIP fraud attempts occur.
My criteria currently are:
To block an IP:
- Minimum 3 attacks within one week from the same IP
To block a network:
- Attacks from minimum 3 IPs from that network within 2 weeks
Common criteria:
- Provider does not react to complaints OR
- Provider sends autoreply
2005 Feb 04
7
Limit MOH processes
You could try to use the native mp3 support for MOH if you really want
mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept
doing nasty things to my system :)
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon
hold.conf there is a section about the native support.
Guillaume
> -----Original Message-----
> From: Stefan Gofferje
2011 Apr 16
4
Jabber / facebook chat?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
- -S
- --
(o_ Stefan Gofferje | SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler & Koch - the original point and click interface
-----BEGIN PGP
2005 Feb 14
1
Flash Operator Panel - lots of problems
On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje
<stefan@gofferje.homelinux.org> wrote:
> Hi folks,
>
> I have some trouble with the FOP and would appreciate if anyone could
> point me into the right direction.
There is a FOP user list, although not too active.
http://www.asternic.org/
> Is there a way to define a button like Zap/g1/6000 and have it light up
> when
2015 Jan 09
2
SEMI OFF-TOPIC - Fail2ban
2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>:
>
> Do you really want to detect "ChallengeSent"? That should occur also on
> legitimate login processes...
>
Hi , strange thing is that I still have not this asterisk in
production and I see many attempts Connection.
Now keep in mind that when a connection of authentication is
successful the
2014 Jun 03
3
Get last dialed number in a context?
Hi,
I would like to implement an auto-redial function in a context. The idea
is about like this:
Dial a number
Hear busy
Hangup
Pick up again
Dial a code like *123
=> jumps into a context which redials until callresult is not busy
Maybe like this:
[autoredial]
exten => s,1,Set(number=${CHANNEL(lastdialed)})
exten => s,2,Dial(SIP/${number}@account,60,g)
exten => s,3,Wait(15)
exten
2011 Apr 16
4
Jabber / GTalk / hints
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com
But the hint doesn't show any difference. It always shows online on the
phone and core show hints always shows that:
6003 at internal : SCCP/6003 State:Unavailable Watchers 0
6002 at internal :
2005 Jan 29
0
SIP Caller ID Number vs. Caller ID Name
Stefan Gofferje wrote:
> Hi folks,
>
> I have a rather nasty problem. I have set up an asterisk test system
> with a Cisco phone, a X-Lite client and so on and did some testing.
> To the developers: great work! Hell of great!
> However, it seems to me like asterisk puts the Caller ID Number into the
> SIP Display Name and Called ID into the Caller ID Number. That is kinda
2005 Feb 16
4
Dutch VOIP-PSTN provider
Hi,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
--
Michiel van Baak
http://lunteren.vanbaak.info
michiel@vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
2005 Feb 08
11
More complicated huntgroups / delayed ringing
Stefan Gofferje wrote:
> Hi Folks,
>
> on my home asterisk, I have a "huntgroup" for incoming calls on the
> private line which first let ring my phones in my office and living
> room, after a while then office, living room and bedroom.
> I do this by simply putting two dial statements in sequence:
>
>
> [private_huntgroup_day]
> exten =>
2005 Jun 25
3
* 1.0.8: no more reacting to callerid?
It's not just you. Same thing happens here. I went back to 1.0.7.
Stefan Gofferje wrote:
> Hi folks,
>
> I used to have some constructions like
>
> exten => number/callerid,1,Goto(somewhere)
>
> After updating to 1.0.8 those does not work any more.
> Any hints?
>
> Regards,
> Stefan
>
2005 Jul 02
2
Colored asterisk -R?
Hi folks,
when I start asterisk directly, I get a colored CLI. When connect to a
already running asterisk with asterisk -R, it's never colored, despite
I'm running both from the same console (tty). Is there a way to force
asterisk -R into color mode?
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security