I can assure you it works. It is important you can set in the [general] section:
match_auth_username=yes
Leandro
2012/1/19 Frank Church voi...@gmail.com:
Does Asterisk permit multiple registrations to the same host?
Each registration has a different username and password
The purpose is for
Hello all,
We are testing Asterisk 1.8.8.1. In the following scenario, peer 54321 dials
12345:
INVITE sip:12345@10.1.1.88 SIP/2.0
Record-Route: sip:10.1.1.86;lr=on;ftag=5ebe58983f6c0c84o3
Via: SIP/2.0/UDP 10.1.1.86;branch=z9hG4bKa6c1.4be79d43.0
Via: SIP/2.0/UDP
The docs also mentioned something about using a port, ie settting
port=x in the sip.conf. Is that also applicable in this case?
On 19 January 2012 08:18, Leandro Dardini ldard...@gmail.com wrote:
I can assure you it works. It is important you can set in the [general]
section:
Hello,
IMHO asterisk acts exactly as it should. How else do you think it should
it prevent sending out the callerid name or num when you set it to prohib?
Asterisk doesnt support the privacy header for outgoing calls so
changing the name and number is the only way to do this. Maybe you could
do
On Wed, Jan 18, 2012 at 01:06:06PM -0600, Shaun Ruffell wrote:
Another thing you can try in the meantime is switch to DAHDI 2.5.0.2
and edit drivers/dahdi/Kbuild to enable dahdi_dummy which will use
the (relatively inefficient for the purposes of conferencing)
highres timers when loaded by
When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote:
Hello,
IMHO asterisk acts exactly as it should. How else do you think it should
it prevent sending out the callerid name or num when you set it to prohib?
This behaviour is new in 1.8, since in 1.6 it work differently (not
In article 4f168fcc.9070...@jttech.se, Johan Wilfer li...@jttech.se wrote:
I'm in the process of replacing an old server with a new one and are
making som changes in the infrastructure, the biggest change in my eyes
is moving from i386 to AMD64 arch. Yesterday I began migrating some
users from
I´m using asterisk 1.6.2.10
We had a problem of high CPU and near the moment it went up we see a
transfer and the following message
chan_local.c: Huh? Local is being asked to answer?
Any idea what can be the issue??
Thanks a lot!!
--
Tony Mountifield wrote:
It may be a stupid question just displaying ignorance on my part, but
why are you using*AMD*64 architecture on an *Intel* processor?
Surely for 64-bit, you should be using x86_64 architecture instead?
From what I've read, AMD came out with the extended instruction
On 01/18/2012 11:14 PM, virendra bhati wrote:
Yes you may used Dialogic card with asterisk. but it's depends on the
requirements too.
I'm not sure what that response is supposed to mean... I can't really
parse it.
If you want to use Dialogic cards with Asterisk, you'll need to contact
On 01/19/2012 05:56 AM, effie mouzeli wrote:
When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote:
Hello,
IMHO asterisk acts exactly as it should. How else do you think it should
it prevent sending out the callerid name or num when you set it to prohib?
This behaviour is new in
Kevin,
Dialogic doesn't provide any soultion as open source. It provides hardware
base cards for making outbond calls. And they used asterisk as backend for
they card application.
On Thu, Jan 19, 2012 at 6:50 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/18/2012 11:14 PM, virendra
Hi,
I just replaced an ancient Asterisk 1.2 server by a new Asterisk 1.8.8.1
server (running on FreeBSD 8.2p4).
I got almost everything working except for voicemail.
In my extensions.conf I have something like:
exten = 5551234,1,Dial(SIP/101, 20)
exten = 5551234,n,Voicemail(1234,su)
Paul Schenkeveld wrote:
exten = 5551234,n,Voicemail(1234,su)
I'm still running 1.4 (slowly configuring a 10 box), but know that when
going from 1.2 to 1.4, it was required to include context for
voicemail. This is how my 1.4 looks:
exten = s,n,Voicemail(${ARG1}@sip|u)
Doug
--
Ben
Hi,
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew the call came from
our system.
But I have a problem with that, not all phone number
try the following
Set(${CALLERID}=722979797 722979797)
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com]
Sent: Thursday, January 19, 2012 8:03 AM
To: Asterisk Users Mailing List
or this
Set(${CALLERID(all)}=722979797 722979797)
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Thursday, January 19, 2012 8:47 AM
To: Asterisk Users
When the date was Thu Jan 19 2012 15:23:04, Kevin P. Fleming wrote:
On 01/19/2012 05:56 AM, effie mouzeli wrote:
When the date was Thu Jan 19 2012 12:12:04, Stefan Schmidt wrote:
Hello,
IMHO asterisk acts exactly as it should. How else do you think it should
it prevent sending out the
Hello,
I configured asterisk in sip.conf like that:
=
register = username:sec...@sipgate.de:5060/number
[sipgate-out]
port=5060
type=friend
insecure=invite
nat=yes
username=username
fromuser=username
fromdomain=sipgate.de
secret=secret
host=sipgate.de
qualify=5000
canreinvite=no
=
On 01/19/2012 07:23 AM, virendra bhati wrote:
Dialogic doesn't provide any soultion as open source. It provides
hardware base cards for making outbond calls. And they used asterisk as
backend for they card application.
Dialogic cards are useful for both outbound *and* inbound calls. There
If in a multi-tenant environment be aware of
https://issues.asterisk.org/jira/browse/ASTERISK-17198 as VMs cannot be
forwarded :(
--
Thanks, Phil
- Original Message -
Paul Schenkeveld wrote:
exten = 5551234,n,Voicemail(1234,su)
I'm still running 1.4 (slowly configuring a 10
Here is a weird problem that I have had several reports on lately.
Customers using our 1.4 based product placing calls to large companies
(two examples - ATT and ComEd, numbers for which I will provide below)
that have auto-attendants based on voice recog get disconnected - by the
remote end -
Hi Kevin,
Is there any possibility of asterisk supported for dialogic cards in
future. does digium has any plan for supporting it?
Thanks
Vinod Dharashive
On Thu, Jan 19, 2012 at 9:02 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/19/2012 07:23 AM, virendra bhati wrote:
On 19-01-12 20:32, Vinod Dharashive wrote:
Hi Kevin,
Is there any possibility of asterisk supported for dialogic
cards in future. does digium has any plan for supporting it?
The Dialogic Diva boards (formerly known as Eicon Diva Server boards)
are supported by Asterisk via chan_capi
2012-01-19 13:27, Doug Lytle skrev:
Tony Mountifield wrote:
It may be a stupid question just displaying ignorance on my part, but
why are you using*AMD*64 architecture on an *Intel* processor?
Surely for 64-bit, you should be using x86_64 architecture instead?
From what I've read, AMD
2012-01-18 20:06, Shaun Ruffell skrev:
That's pretty severe, and could certainly cause problems for DAHDI
trying to use the kernel as a timing source. NTP will correct the
drift, but the drift is still happening and it's not corrected on
every tick. If the ticks are not happening at the rate
After upgrade to Asterisk 1.8 one of my phones connected to AudoCodes box is
losing SIP registration.
I can call out but when I tray to call that extension it ring busy.
Other times when I call this extension the phone rings and when I hang up the
phone still keep ringing, it does not
On 01/19/2012 01:32 PM, Vinod Dharashive wrote:
Is there any possibility of asterisk supported for dialogic
cards in future. does digium has any plan for supporting it?
Digium's plans for Asterisk, in general, are public knowledge. We have
not published any intent to produce support
2012-01-18 19:44, John Knight skrev:
Have you used 64 bit kernels (amd64) in your setup? Distribution?
Aye, I use the current stable 64-bit rhel6 branch openvz kernel with
centos 6 on the node and scientific linux 6 in the template without
issue other than what I described before with
Often, when I want to be able to do post-mortem analysis of network
traffic, I can have a suitable tcpdump with -w to capture raw packets
for later analysis with Wireshark. On some systems I have this running
continuously on the SIP port.
Is there any way of doing something similar with PRI ISDN
On 01/19/2012 05:25 PM, Tony Mountifield wrote:
Often, when I want to be able to do post-mortem analysis of network
traffic, I can have a suitable tcpdump with -w to capture raw packets
for later analysis with Wireshark. On some systems I have this running
continuously on the SIP port.
Is there
Asterisk Project Security Advisory - AST-2012-001
++
| Product| Asterisk|
I've been quite happy with Debian.
Previously I was using BSD, and it was almost impossible to
upgrade the system. And apt / dpkg have never failed me, very
impressive. I guess rhel works well also, but I've little
experience with it.
The Asterisk Development Team has announced security releases for Asterisk 1.8
and 10. The available security releases are released as versions 1.8.8.2 and
10.0.1. Please note that the security vulnerability in Asterisk 1.8 and 10
does not exist for Asterisk versions 1.4 or 1.6.2.
These releases
I have a honey pot box with extensions that are not just numbers ie )
100-MySipUserName
And the passwords are from an openssl generated password ie)
Gq5VNIjDFWIQoUT6
However, this one extension keeps getting hacked and showing up on a different
IP address.
It is also register
It's funny. The link
Links | https://issues.asterisk.org/jira/browse/ASTERISK-19202
Produces:
Permission Violation
It seems that you have tried to perform an operation which you are not
permitted to perform.
If you think this message is wrong, please consult your administrators
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