* You are viewing Posts Tagged ‘call’

Group call from DAHDI

Hello,

I am trying to figure out how to make group call using DAHDI. I want to
make multiple call at once and conference among them. I know about meetme,
but that is for incoming conference call.

Please suggest.

Remote crash vulnerability in IAX2 channel driver.

Asterisk Project Security Advisory – AST-2012-007

Product Asterisk
Summary Remote crash vulnerability in IAX2 channel driver.
Nature of Advisory Remote crash
Susceptibility Established calls
Severity Moderate
Exploits Known No
Reported On March 21, 2012
Reported By mgrobecker
Posted On May 29, 2012
Last Updated On May 29, 2012
Advisory Contact Richard Mudgett < rmudgett AT digium DOT com >
CVE Name CVE-2012-2947

Description A remotely exploitable crash vulnerability exists in the
IAX2 channel driver if an established call is placed on
hold without a suggested music class. For this to occur,
the following must take place:

1. The setting mohinterpret=passthrough must be set on the
end placing the call on hold.

2. A call must be established.

3. The call is placed on hold without a suggested
music-on-hold class name.

When these conditions are true, Asterisk will attempt to
use an invalid pointer to a music-on-hold class name. Use
of the invalid pointer will either cause a crash or the
music-on-hold class name will be garbage.

Resolution Asterisk now sets the extra data parameter to null if the
received control frame does not have any extra data.

Affected Versions
Product Release Series
Certified Asterisk 1.8.11-cert All versions
Asterisk Open Source 1.8.x All versions
Asterisk Open Source 10.x All versions

Corrected In
Product Release
Certified Asterisk 1.8.11-cert2
Asterisk Open Source 1.8.12.1, 10.4.1

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2012-007-1.8.11-cert.diff v1.8.11-cert
http://downloads.asterisk.org/pub/security/AST-2012-007-1.8.diff v1.8
http://downloads.asterisk.org/pub/security/AST-2012-007-10.diff v10

Links https://issues.asterisk.org/jira/browse/ASTERISK-19597

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2012-007.pdf and
http://downloads.digium.com/pub/security/AST-2012-007.html

Revision History
Date Editor Revisions Made
05/29/2012 Richard Mudgett Initial release.

Asterisk Project Security Advisory – AST-2012-007
Copyright (c) 2012 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

Call Forwarding

Hi Guys,

Seeing an issue with 1.6.2.17.2 and also 1.6.2.14

When we do call forwarding if the call coming in to be forwarded
asterisk sends the invite out to our ITSP as
username@anonymous.invalid instead of username@domain.

When call comes in with CLI and is forwarded it sends it as
username@domain to our ITSP.

Is this a bug or is there something I need to turn on or off? All the
ITSP’s we use authenticate on username and domain.

Thanks
Brian.

Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’

On Wed, May 23, 2012 at 02:02:51PM +0500, p070075 Muhammad Atif Ramzan wrote:
> Hi
>
> Can anyone help me with this error
> Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’
>
> i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
> reached the destination but no voice is coming from destination my voice
> reflects back

Have you verified the user asterisk is running as can execute
dahdi_scan? This was asked not too long ago on the forums as well:

http://forums.asterisk.org/viewtopic.php?f=1&t=82659

twenty thousands (20, 000) users, which asterisk and how many servers

the solution lies in kamailio/opensips’s despatcher module.

Sent from my iPhone

On 23 maj 2012, at 20:46, bilal ghayyad wrote:

> Dear;
>
> So it is a hardware issue and not software?
> I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls.
>
> About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server?
>
> Regards
> Bilal
>
> ————–
>>
>> 20.000 users is really a big number, as big as 2000
>> concurrent calls.
>> As previously stated on this list, it depends… it depends
>> by the type of
>> calls for example. If all media is offloaded from the server
>> letting the
>> phones to reinvite each other, than your server CAN support
>> the call
>> volume. If instead even a tiny portion of the call volume
>> uses service on
>> the pbx, like IVR, music on hold, conferences, queues or
>> even worst,
>> transcoding, then the server is obviously underpowered. From
>> my point of
>> view, servicing 20.000 users with a single piece of hardware
>> is highly
>> risky. It can broke in the middle of the day, leaving all
>> your users
>> without service. I think a better approach will be to have
>> more less
>> powered servers working all together to serving your users.
>> If a day one or
>> two of them broke, you have not to worry because the other
>> will continue to
>> serve your users and nobody notice the little decrease in
>> power.
>> There are a lots of way to achieve the high availability,
>> load sharing,
>> each with its pros and cons.
>> Right now I am building a pbx with high availability and
>> load sharing in
>> mind, for a client who wants to achieve numbers you have
>> just said. Let’s
>> see how it works in few months.
>>
>> Leandro
>>
>> 2012/5/23 bilal ghayyad
>>
>>> Hi All;
>>>
>>> I need to use Asterisk for 20 000 users, so which
>> asterisk version to be
>>> used? Is there asterisk version that supports 20,000
>> users on one hardware
>>> machine?
>>>
>>> Can I use one strong hardware server i7 with 64 GB RAM
>> and fast hard desk
>>> to handle 20 000 users, and concurrent calls 2000? Or I
>> need multiple
>>> servers, how much?
>>>
>>> If I am going to use multiple servers (until now I do
>> not know how much,
>>> and I do not know if the barrier will be the asterisk
>> software or the
>>> hardware), then do I have to use special SIP proxy or I
>> have to use load
>>> balancer)? In this case, I have to use asterisk
>> Database (so all the
>>> servers will read/write from the database)?
>>>
>>> What about AsteriskNow, can it support?
>>>
>>> Regards
>>> Bilal
>
> –
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Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’

Hi

Can anyone help me with this error
Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’

i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
reached the destination but no voice is coming from destination my voice
reflects back

thanks