Hi Daniel/asterisk users,
You're correct, a typo.
If got now to stream configured in musiconhold.conf
[Hitz]
mode=custom
application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
http://scfire-dtc-aa02.stream.aol.com:80/stream/1074
[sbs]
mode=custom
application=/usr/local/bin/mpg123
On Mon, 27 Dec 2010, Asim Amin wrote:
Also since some of these manufacture only analog cards,
does anyone have any experience using these in a single system with digital
cards from other manufacturers like Openvox?
I've used OpenVox analogue cards. They seem to just work without having
to do
On Mon, 27 Dec 2010, Olivier wrote:
2010/12/25 dave george dgeo...@teletoneinc.com
Need some advise or paid help on running asterisk on two WAN connection. I
need load balancing and failover support.
WAN: 1 DSL + 1 Cable ISP.
I seem to have missed the start of this - however I'd suggest
The biggest issue with any solution to use two different providers for
your IP service that will be used by your VOIP provider to deliver
calls to your Asterisk server, is that each internet service will have
a separate address. Therefore, for INBOUND calls, your VOIP provider
will have to do the
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote:
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp to match fail2ban requirements.
How about posting your fail2ban config?
--
Daniel Tryba
--
On Mon, Dec 27, 2010 at 09:56:56AM +0100, Arjan Kroon | Mobillion wrote:
[sbs]
mode=custom
application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx
The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs
The
On Mon, Dec 27, 2010 at 3:38 AM, Olivier oza_4...@yahoo.fr wrote:
2010/12/26 Richard Kenner ken...@gnat.com
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from
LDAP.
Can this be done? If so, with
I need clarification on couple of issues of Realtime Queue.
It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk
puts this Queue-Member relationship information into AstDB, So that on
asterisk restart this can be preserved.
My question is, why does asterisk not store
On Mon, 27 Dec 2010 09:14:22 + (GMT), Gordon Henderson
gordon+aster...@drogon.net wrote:
I've used OpenVox analogue cards. They seem to just work without having
to do anything special.
+1. I have an OpenVox with a single FXO module, and it's been working
for 4 years now. I don't know the
some providers do serve inbound by sending the traffic to exact IP, some do
accept the registers from any IP.
in second case for Inbound failover, you might just to register = using
another interface/IP address.
here a new question arose: how to sip-ping some phone number to see if
it's alive?
Hello!
I am wondering how the differences between G729, G729a, and G729b
effect call bridging and server interoperability. For example, can
one server use the G729 code with another server that uses the G729A
codec?
Also, which version is Asterisk set up to use?
Thanks!
Elliot
--
On Wednesday, December 22, 2010 04:59:42 pm Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan
Kayhan Sent: Wednesday, December 22, 2010 4:11 AM
To: 'Asterisk Users Mailing List -
Hi,
I wonder what conditions might lead, that SIP packets from provider
P destined to my external SIP server A, are reaching my internal SIP server
B?
the fun factor is that internal B server is used for outbound calls via the
same provider P.
I found no routing issues.
Is it possible to
We have another gateway in the USA that will send traffic to both IPs. The
US gateway will load balance the traffic to both IPs.
This is not used for phones. It is used mainly for wholesale traffic.
Asterisk is being used as an SS7 gateway.
Each DSL limits us to about 16 calls. We are
Hi,
Have you checked SIP messages on B server? Maybe your provider P
sends traffic to incorrect IP.
--
Best Regards,
Giedrius
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Surely. B responds 404 Not Found., as it's not configured to receive these
SIP packets.
provider P sends to correct IP, and moreover B has no external IP.
On Mon, Dec 27, 2010 at 3:54 PM, voipas voi...@gmail.com wrote:
Hi,
Have you checked SIP messages on B server? Maybe your provider P
jail.conf
[asterisk-iptables]
enabled = true
filter = asterisk
action = iptables-allports[name=ASTERISK, protocol=all]
sendmail-whois[name=ASTERISK, dest=root,
sender=fail2...@example.org]
logpath = /var/log/asterisk/messages
maxretry = 5
bantime = 259200
filter asterisk.conf
- Original Message -
Anyone who has experience using Digium analog card clones from any of the
following:
1. Zycoo
2. CTVON
3. Chinaroby
4. Etross
5. Immediate IT (IIT)
6. Realtone
and can give review which one is good quality with easy configuration and
error free running.
On 12/27/2010 08:05 PM, Elliot Murdock wrote:
Hello!
I am wondering how the differences between G729, G729a, and G729b
effect call bridging and server interoperability. For example, can
one server use the G729 code with another server that uses the G729A
codec?
Also, which version is Asterisk
Le 27/12/2010 16:20, dave george a écrit :
[...]
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST
can
# be used for standard
Simply to reduce the attack, and then improve the defense:
If you don't need traffic from some area that is attacking you, just put the
whole area in IPTables. A list is available on VOIP-INFO.org.
Cull out what you want to allow.
Then tune Fail2Ban at your leisure.
Cary Fitch
--
With asterisk 1.8+ it should be:
failregex = NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Wrong password
NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Monday, December 27, 2010 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] live
On Mon, Dec 27, 2010 at 10:20:13AM -0500, dave george wrote:
[snip fail2ban config]
Well, all looks fine. Your filter is correct. Your message log is also in the
correct format. You can test this with:
fail2ban-regex /var/log/asterisk/messages /etc/fail2ban/filter.d/asterisk.conf
So is fail2ban
Dnia Sat, 25 Dec 2010 15:31:57 +0200
Michael voip.quest...@gmail.com napisał(a):
Is that possible?? From what we saw, the agents login works on a
constantly open line.
Which version of Asterisk you're using?
--
Damian Ryszka aka Rychu
rychu(at)sileman.net.pl
--
I am setting up CEL with asterisk 1.8 and so far so good. The issue I was
hoping to address here was also being able to get storage of other values
such as HANGUPCAUSE and other variables that are used for billing and
quality of service. The CEL documentation starts out by saying that we can
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a bit
further and use it at cmmand level to be able to send SIP notifies to
restart a phone or take advantage of a phone's UPnP capabilities. Is
Asterisk capable of that? If so, what is a simple SIP reboot message
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Monday, December 27, 2010 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Using SIP stack within Asterisk to reboot
On 12/27/2010 12:08 PM, Bruce B wrote:
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a
bit further and use it at cmmand level to be able to send SIP notifies
to restart a phone or take advantage of a phone's UPnP capabilities. Is
Asterisk capable of that?
Hi,
we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to
problems with iax channel posted earlier, we wanted to switch back to
1.4 version.
Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is
recognized and the 7 euroISDN channels are running well,
On Mon, Dec 27, 2010 at 3:08 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a bit
further and use it at cmmand level to be able to send SIP notifies to
restart a phone or take advantage of a phone's UPnP capabilities.
On Mon, Dec 27, 2010 at 3:33 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 12/27/2010 12:08 PM, Bruce B wrote:
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a
bit further and use it at cmmand level to be able to send SIP notifies
to restart a
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote:
Hi,
we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to
problems with iax channel posted earlier, we wanted to switch back to
1.4 version.
Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is
recognized
Lots of good info and pointers so far. But do keep in mind that not all
phones will automatically reboot just because you sent it a check-sync or
resync event with the sip notify command.
I vaguely remember that for e.g. the Polycoms some other condition had to be
true: either the phone's config
Thanks Kai-Uwe and everyone else. I have seen all those examples and I am
exploring the sip_notify.conf file now which makes things more clear to me.
However, when sending a SIP notify to a phone that is not registered to
Asterisk via it's IP address should I expect to receive a success of fail
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Monday, December 27, 2010 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using SIP stack within Asterisk to reboot
phones -
On Mon, 27 Dec 2010, Bruce B wrote:
Thanks Kai-Uwe and everyone else. I have seen all those examples and I
am exploring the sip_notify.conf file now which makes things more clear
to me. However, when sending a SIP notify to a phone that is not
registered to Asterisk via it's IP address should
Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0
System with asterisk ?
I run a small asterisk server at home using two SPA3102s, and thinking of
upgrading my cordless analog phones to something a little newer.
--
I've never worked with Aastras, so don't have any additional data over
what's been said by others. Also, I've never sent the SIP check-sync
notify to a phone that wasn't already registered with the asterisk server
the SIP notify was sent from. My best *guess* would be that actual behavior
of the
I have no direct experience. But I know that E4 Technologies has been
using this phone with Asterisk Switchvox. Panasonic made an effort
earlier this year to have it certified with Asterisk. It's also
Broadvoice certified.
Michael
--Original Message Text---
From: William Stillwell
Date: Mon, 27
No, I don't know how to do this. Does anybody?
I'd like to take a voicemail file from asterisk (*.wav, *.gsm, *.mp3 ?)
and send it to googlevoice as a voicemail, then get the transcription
over gmail.
I know about pygooglevoice (is it still maintained?). But I can't figure
out how to dial
SIPp is a good option.
Thanks
Nikhil
On 12/27/2010 11:38 PM, Bruce B wrote:
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it
a bit further and use it at cmmand level to be able to send SIP
notifies to restart a phone or take advantage of a phone's UPnP
What type of phones? Easy to do with Polycom and several others from Asterisk
CLI.
Sent from my BlackBerry® smartphone
-Original Message-
From: Nikhil d.nik...@cem-solutions.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 28 Dec 2010 08:42:22
To:
Hi,
I have used 4-PRI card from atcom.cn and it works perfectly for me.
Regards,
Faisal
+923214059996
On 12/27/2010 12:25 PM, Asim Amin wrote:
Hello All,
Anyone who has experience using Digium analog card clones from any of
the following:
1. Zycoo
2. CTVON
3. Chinaroby
4. Etross
5.
Hi,
We're using version 1.6.2.X.
I think that the command we need is AgentCallbackLogin. We're building a
script to study the entire functionality of queues, agents and everything
around it.
Happy New Year to all,
Michael
2010/12/27 Damian Ryszka ry...@sileman.net.pl
Dnia Sat, 25 Dec 2010
Hi Everyone,
I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can
originate calls see the program login nicely but when a call comes in it
only shows the Name portion of the CLID and not the number hence it pulls up
a new contact on Outlook. The new contact only show name
Dnia Tue, 28 Dec 2010 08:02:51 +0200
Michael voip.quest...@gmail.com napisał(a):
I think that the command we need is AgentCallbackLogin. We're
building a script to study the entire functionality of queues,
agents and everything around it.
Perhaps you noticed, that AgentCallbackLogin() has
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