Hey Josue,
Thanks alot. I will be expecting the configuration samples. From your
response, I guess QSIG would be better for more functionality between the
two PBXs then..
Yes, this is my first implementation of asterisk and the support I have had
from the mailing lists (some just by searching
Hello list,
is it possible to extract the Remote-Party-ID from an incoming call in
the dialplan ? Is there some kind of function for this ?
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by
Hi Users,
Does Asterisk provide any way to monitor the SIP call setup time between the
clients ??
I understand that there is a way to monitor the RTP data flow for jitter and
packet losses
using *ship show channelstats*. I am looking something on similar lines to
monitor call
setup time during
On Tue, 15 Mar 2011 13:45:00 -0400, Paul Belanger
pabelan...@digium.com wrote:
Is this an analog line? If so, is your CO providing a disconnect tone?
Yes, it's an analog line, but it's actually VoIP provided by an RJ11
on an ADSL modem, not a real landline.
Is there a way to check how the
Here is a better link for DUNDi
http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
skip the part which you know already
On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.comwrote:
[]'sf.rique
On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger
On Mon, Mar 14, 2011 at 07:19:18PM -, Paddy Grice wrote:
Hi List
I am having trouble running the command
siptest:~# asterisk -rx 'dialplan reload'
most times it does what I expect and I get a response as below
siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded.
Hi
Does anyone know what this error is about?
I've had 0 success in trying to find any reference to it on the internet
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
--
Hi There,
i have the same problem but it doesnt always happen tho from the same
caller.
im using Asterisk 1.4 - maybe newer version updates have
had bug fixes. maybe this could rectify it.
Regards,
On Tue, 2011-03-15 at 14:54 +0100, Gilles wrote:
Hello
I'm trying to use
Ishfaq Just wondering if that Server xxx.xxx.xxx.xxx has firewall rules
that are blocking ports for TCP and UDP.
Kind Regards,
John.
On Wed, 2011-03-16 at 11:09 +, Ishfaq Malik wrote:
Hi
Does anyone know what this error is about?
I've had 0 success in trying to find any reference
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, March 16, 2011 2:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract Remote-Party-ID from incoming
Kevin P. Fleming kpflem...@digium.com writes:
Why do you need a Local channel to do this? If extension 234 exists in
some context, the Dial() statement in that extension can dial
SIP/234-foo and SIP/234-bar itself.
Good point.
It can be a bit of fun keeping track of the phones when they are
From: satish patel satish...@hotmail.com
Sent: Tuesday, March 15, 2011 2:31 PM
To: asterisk-users asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call file for page auto-call
Thanks for you input but how to do SIPAddHeader(Alert-Info:
Hello.
I would need some help trying to setup Asterisk 1.6.2.9-2+squeeze1
on a Debian 6.0 system. I'd like to use the Debian packages, hence
the strange version number…
Since I'm new to Asterisk, I'm trying to follow The Asterisk Book
at
16 mar 2011 kl. 14.13 skrev Benny Amorsen:
Kevin P. Fleming kpflem...@digium.com writes:
Why do you need a Local channel to do this? If extension 234 exists in
some context, the Dial() statement in that extension can dial
SIP/234-foo and SIP/234-bar itself.
Good point.
It can be a
Hi,
Here is a scenario:
1) A call comes in on an outside line on a DAHDI device
2) The call is answered by a SIP extension (Linksys SPA942 to be exact)
3) The SIP extension places the outside call on hold
4) The same SIP extension dials another extension.
Is it possible for the dialplan in
Never mind. Its in seconds :)
On Tue, Mar 15, 2011 at 6:48 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi all,
What is the unit of asterisk AMI events timestamp value?
milli/micro etc ?
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E:
On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
is it possible to extract the Remote-Party-ID from an incoming call in the
dialplan ? Is there some kind of function for this ?
Kind regards,
Jonas.
1.8 Documentation on Connected Line update. Works like magic.
On 03/16/2011 02:56 PM, Andrew Latham wrote:
On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
is it possible to extract the Remote-Party-ID from an incoming call in the
dialplan ? Is there some kind of function for this ?
Kind regards,
Jonas.
1.8
Hi,
I'm in a hosted PBX context. I'd like to push some strings (Hello World)
from Asterisk to a Polycom phone's screen that is behind a NAT/firewall.
Basically as part of the SIP if possible, to make it firewall friendly.
Is this even possible?
Part 2: Is this possible as part of a
I'm in a hosted PBX context. I'd like to push some strings (Hello World)
from Asterisk to a Polycom phone's screen that is behind a NAT/firewall.
Basically as part of the SIP if possible, to make it firewall friendly.
Is this even possible?
Part 2: Is this possible as part of a queue
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, March 16, 2011 10:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Pushing info to a Polycom phone - from outside
Hi,
I got actually an * connected to HiPath 3750 via a trunk of 2xE1, here
is my dahdi config :
/etc/asterisk/chan_dahdi.conf
[channels]
language=fr_FR
usecallerid=yes
hidecallerid=no
callerid=asreceived
restrictcid=no
usecallingpres=yes
pridialplan=unknown
prilocaldialplan=dynamic
I'm in a hosted PBX context. I'd like to push some strings (Hello World)
from Asterisk to a Polycom phone's screen that is behind a NAT/firewall.
Basically as part of the SIP if possible, to make it firewall friendly.
Is this even possible?
Part 2: Is this possible as part of a queue
Den 19-01-2011 00:19, Nick Ustinov skrev:
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info
[2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer
I'm trying to run a shell command from AMI, but I guess I'm doing something
wrong or there's a bug because no matter what command I try I always get a null
response. Running the latest 1.6.2 release.
On manager.conf I have:
[test]
secret = test
deny = 0.0.0.0/0.0.0.0
On Wednesday 16 March 2011 06:09:33 Ishfaq Malik wrote:
Does anyone know what this error is about?
I've had 0 success in trying to find any reference to it on the internet
Well, the most obvious problem is that you cannot send (or bind, or do
anything, really) to port 0.
--
Tilghman
--
On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote:
action: command
command: ! /bin/ls -l /
For security reasons, you cannot do this. This is intentional, not a bug.
Consider the command 'rm -rf /' for the reason why.
--
Tilghman
--
- Mensagem original -
On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote:
action: command
command: ! /bin/ls -l /
For security reasons, you cannot do this. This is intentional, not a bug.
Consider the command 'rm -rf /' for the reason why.
--
Tilghman
I understand the
Hello,
When I want to send a call from asterisk-server 1 to asterisk-server 2,
it fails.
On Asterisk server 1 :
register = user:passwd@asterisk1 ; Test TRUNK
[trunk2]
type=peer
host=asterisk1
username=user
;defaultuser=user
secret=passwd
disallow=all
allow=alaw
allow=gsm
qualify=yes
On 03/16/2011 08:39 PM, Jonas Kellens wrote:
On Asterisk server 2 I see the following when I make a call with a
Grandstream IP-phone, registered at Asterisk server 1 :
[Mar 16 20:32:44] WARNING[1680]: chan_sip.c:12872 check_auth: username
mismatch, have test7, digest has user
[Mar 16
On Wednesday 16 March 2011 14:11:21 Vinícius Fontes wrote:
I understand the concern with security but why not create a separate
authorization allowing that instead of hard-coding it?
I understand the concern with security but why not create a separate
authorization allowing that instead of
I am reading about, and some people are saying that openser is better for
biger envoriments, and dundi is fine for smal envoriments, does anyone have
any info about it ?
We have now about 4500 convencional phones and we gonna expand a lot.
So,
OpenSER vs DUNDi ?
I guess i will use Asterisk
- Mensagem original -
On Wednesday 16 March 2011 14:11:21 Vinícius Fontes wrote:
I understand the concern with security but why not create a separate
authorization allowing that instead of hard-coding it?
I understand the concern with security but why not create a separate
On 16/03/11 5:43 PM, Nikhil wrote:
ok..that means I have to modify chan_sip . I wondering why this is not
available in asterisk.
Because you haven't completed the patch yet! :P
--
Cheers,
Matt Riddell
___
http://www.venturevoip.com/news.php (Daily
Well, it has disappeared in further builds ;)
Thanks
2011/3/16 Leif Neland le...@neland.dk:
Den 19-01-2011 00:19, Nick Ustinov skrev:
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624]
On 17/03/11 9:53 AM, Vinícius Fontes wrote:
No increased security, lots of hassle, all because there's an
undocumented feature that is supposed to increase security but just
takes functionality away.
If you really want to you could add some dialplan like:
[dangerous]
exten =
But what about if asterisk running with non-privilege user?
Still it is not a good idea.
--
Sent from my iPhone
On Mar 16, 2011, at 2:33 PM, Tilghman Lesher tilgh...@meg.abyt.es
wrote:
On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote:
action: command
command: ! /bin/ls -l /
- Mensagem original -
On 17/03/11 9:53 AM, Vinícius Fontes wrote:
No increased security, lots of hassle, all because there's an
undocumented feature that is supposed to increase security but just
takes functionality away.
If you really want to you could add some dialplan like:
But what about if asterisk running with non-privilege user?
Still it is not a good idea.
Yes I forgot to say that I also run Asterisk as a regular user, which also
helps with security.
But I really don't see much of a threat on this. AGI does almost the same. --
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinícius
Fontes
Sent: Wednesday, March 16, 2011 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Executing shell commands via AMI
Hi,
I've been trying to set up multiple parking lots using multiple tenants on
version 1.8.x (tried all versions including 1.8.4RC2), however calls only
park on one parking lot (the top parking lot of the command 'parkedcalls
show').
Everything works fine when running on version 1.6.2.17.
If you want total control from AMI then point at an extension that you can set
variables to commands and arguments, call an AGI and set variables that can be
passed back to AMI via user events.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 16, 2011, at 3:03 PM,
ProductAsterisk
SummaryResource exhaustion in Asterisk Manager Interface
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions if
ProductAsterisk
SummaryRemote crash vulnerability in TCP/TLS server
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions
The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.
These releases are available for immediate download at
On Wed, 16 Mar 2011, Vinícius Fontes wrote:
But I really don't see much of a threat on this. AGI does almost the same.
I thought you didn't want to start a flamefest :)
The security risk of AGI would be 'the same' if you provide a method for a
miscreant to create a file on your Asterisk
Hi, I am having a little problem and I hoped maybe I could get some help
here.
I deployed a Asterisk 1.8 server of my own to make SIP calls just between my
friends. The server is configured with a public IP address.
My friends and I are using Acrobits Softphone for iPhone as a client.
I am using
I listened to your email using DriveCarefully and will respond as soon as I can.
Download DriveCarefully for free at www.drivecarefully.com
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Sent from my BlackBerry®
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On Wed, Mar 16, 2011 at 9:07 PM, edward choi mp2...@gmail.com wrote:
Now, the current situation is like this:
My friend is under a WI-FI access point at his home, so his iPhone is
assigned something like 192.168.x.x.
I am using 3G network, so I have a public IP address.
snip
I just don't
Thanks for the info.
But then do I have to set 'nat=no' when he is on a public IP address?
It would be quite a labor to switch back and forth every time my friend
switches from a public to private IP or private to public IP.
2011/3/17 Warren Selby wcse...@selbytech.com
On Wed, Mar 16, 2011 at
On Wed, Mar 16, 2011 at 11:39 PM, edward choi mp2...@gmail.com wrote:
Thanks for the info.
But then do I have to set 'nat=no' when he is on a public IP address?
It would be quite a labor to switch back and forth every time my friend
switches from a public to private IP or private to public
ED, doesnt matter whether your using Public or Private IP addresses it
should still work. theres also a situation on how you have configured
it. it can also be a codec issue. i havent really dealt with Acrobits
i would check the codec's if its using GSM or G.711 which is standard.
and also check
On Wed, Mar 16, 2011 at 11:41 PM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Mar 16, 2011 at 11:39 PM, edward choi mp2...@gmail.com wrote:
Thanks for the info.
But then do I have to set 'nat=no' when he is on a public IP address?
It would be quite a labor to switch back and forth every
Ok, I did a little test.
First, I set 'nat=yes' in my account and my friend's account in sip.conf.
Now when we are both on 3G, we can hear our voices just fine.
When we are under the same WI-FI access point(ie private IP address), we can
hear our voices fine.
BUT, when I am on 3G, and my friend
Dear mailing list,
I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian
and I've a strange behavior.
After some days running normally, my asterisk is under heavy attack,
however, there is nothing logged in the console (logging from debug -
error) or file (level from notice
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