Dnia Tue, 28 Dec 2010 08:02:51 +0200
Michael 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 been removed in 1.6
seri
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 and
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
> Dnia Sat, 25 Dec 2010 15:31:57 +0200
> Mich
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. Immed
What type of phones? Easy to do with Polycom and several others from Asterisk
CLI.
Sent from my BlackBerry® smartphone
-Original Message-
From: Nikhil
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 28 Dec 2010 08:42:22
To:
Reply-To: Asterisk Users Mailing List - Non-Comme
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
capa
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 g
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
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
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.
--
_
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
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 - Po
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
pa
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 and
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 co
On Mon, Dec 27, 2010 at 3:33 PM, Kevin P. Fleming 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 phone or tak
On Mon, Dec 27, 2010 at 3:08 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 ca
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, ingoin
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? If
_
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 pho
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
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
no
Dnia Sat, 25 Dec 2010 15:31:57 +0200
Michael 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
--
__
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 a
-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 audio
With asterisk 1.8+ it should be:
failregex = NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5})?' - Wrong password
NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5})?' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5
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
--
__
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 ""
can
# be used for standard IP/hostnam
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
- 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 r
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
[I
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 wrote:
>
> Hi,
>
> Have you checked SIP messages on B server? Maybe your provider P
> sends tr
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
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 thinki
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 buil
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 Li
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
--
_
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 al
On Mon, 27 Dec 2010 09:14:22 + (GMT), Gordon Henderson
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 other manufacturers listed
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 call
On Mon, Dec 27, 2010 at 3:38 AM, Olivier wrote:
> 2010/12/26 Richard Kenner
>> 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 what Asterisk versions?
>
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
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
--
_
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 l
On Mon, 27 Dec 2010, Olivier wrote:
2010/12/25 dave george
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 getting
hardware to do i
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 d
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
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