bayardo.sanc...@gmail.com wrote:
This week I was experiencing attacks sip log into my accounts were more than
1,000 requests for records Sip accounts in less than an hour THROUGH deny
the ip of my router access list in cisco and my asterisk server to go through
the iptables drop ip
What happens if you put in a 'room' number?
Eg: exten = 8080,3,MeetMe(500|MDci)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
touati
Sent: 17 September 2010 14:24
To: Asterisk Users Mailing List -
Hello list,
I read this in sip.conf :
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
What does this mean ?!
Does this mean that when I mark this as yes, a phone that already has
taken a call will be
Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This setting is only
Hi,
I use asterisk with sip3000 device with sip-aho connected to PSTN and
sip-ahi connected to a phone.
When call arrives from PSTN, the *phone continues ringing even after caller
hanged up*.
The dialplan contains the following lines:
[from-pstn]
...
exten = 99,n,Dial(SIP/sip-ahi,30,g)
exten =
Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This
On 20 September 2010 05:33, dashy dude dashy_v2...@yahoo.com wrote:
Hi,
I tried disabling cdr_addon_mysql.so.
Still error comes let's say once a day or so.
Is there anything else I can do about?
rgds
--- On Thu, 9/9/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de
Hi
I have a call established and I want to play audio to just one channel
on that call. Is this possible? If so, how? My google-fu has failed on
this one.
Regards
Jon
--
Jon Farmer
Tel 07795 118140
--
_
-- Bandwidth and
it's going to put you in conf no 500 without prompting you to enter a
conference number I guess, but i don't it's going to solve my issue.
actually I'm atill wondering is there a way to debug just Meetme app output
or the only way is turn the whole debug thing on?
On Mon, Sep 20, 2010 at 4:11 AM,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Monday, September 20, 2010 7:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Playing Audio To One
One way to do it is to use ChanSpy and the whisper option. We use AMI to play
sound bits to one leg of the call.
Something like
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable:
On 20 September 2010 14:23, Danny Nicholas da...@debsinc.com wrote:
One option would be to play your audio through a conference; Asterisk seems
to have great controls over legs using that infrastructure.
That is not an option. I am using Asterisk as a media relay and want
to play a message
Have you tried removing option 'g' from your Dial command?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote:
Hi,
I use asterisk with sip3000 device with sip-aho connected to PSTN and
sip-ahi connected to a phone.
When call arrives from
On 20 September 2010 14:21, Jim Dickenson dicken...@cfmc.com wrote:
One way to do it is to use ChanSpy and the whisper option. We use AMI to play
sound bits to one leg of the call.
Something like
Hi
I have tried your suggestion however I can't get it to work. When I
send the originate via
Hi,
On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote:
Have you tried removing option 'g' from your Dial command?
Of course, with the same result.
--
Arie
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote:
Check the SIP debug and see what is going on.
Leif.
Hi,
I checked the SIP debug.
As soon as I issue the RELOAD command, no SIP data gets transferred to the
phone.
Asterisk output: http://pastebin.com/FB675N16
Any ideas how I can do a SIP reload without losing the Sip Phones registration?
Anyone have a AudioCodes with Asterisk ???
2010/9/18 Olivier CALVANO o.calv...@gmail.com:
Hi
i have buy a Audiocode Median 2000 VoIP Gateway and connect it on :
1 E1 30 channels
1 Lan Port
Anyone use this equipements with asterisk ? because i am search a
config sample
Do you mean spa3000 or sip3000? I remember having same problem with spa3000
and the problem was somewhere in the settings of spa3000 that wouldn't stop
ringing the phone. I don't remember the details at this moment as it was
long time ago, but this much I can tell that it is a config issue with
I am not aware of any way to do that. My question is if you are using
realtime, why are you doing a sip reload? If you change the settings on a
device in the realtime DB, just prune it and it will grab the new config the
next time they re-register.
-Original Message-
From:
Hello List,
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
Can we not do pastebin any more?
I just received this:-
[PASTEBIN URL REMOVED] has been detected as suspicious URLs,and Quarantine to
user's spam folder has been taken on 9/20/2010 8:24:38 AM.
Message details:
Server: MADRID
Sender: d...@keshrcommunications.com;
Recipient:
On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Anyone have a AudioCodes with Asterisk ???
Yes, but why? Both do the same thing. It would be like me asking 'I
have a bike and need to get to work. Can I use the bike with a car?'
--
Paul Belanger | dCAP
On Mon, Sep 20, 2010 at 7:31 AM, Arie Skliarouk sklia...@gmail.com wrote:
When call arrives from PSTN, the phone continues ringing even after caller
hanged up.
I suspect a bug [1] but without a SIP debug, I cannot be sure.
[1] https://reviewboard.asterisk.org/r/870/
--
Paul Belanger | dCAP
Last week I had a couple of outages one machine, the problem was that
Asterisk suddly stopped responding to UDP SIP requests. tcpdump show
requests arriving on the machine, sip debug log in asterisk doesn't show
anything for the UDP peers, TCP functions just fine.
In all 3 cases the log is
My question is if you are using realtime, why are you doing a sip reload?
I said previously:-
Let's say I add a new provider to my service and therefore have to add
another register= command into sip.conf, I'd have to issue a sip reload
which would kill off all the realtime sip phones.
- Paul Belanger paul.belan...@polybeacon.com wrote:
On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO
o.calv...@gmail.com wrote:
Anyone have a AudioCodes with Asterisk ???
Yes, but why? Both do the same thing. It would be like me asking 'I
have a bike and need to get to work.
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote:
Can we not do pastebin any more?
No, it's just one user with an excessively paranoid and chatty
mailfilter.
--
_
-- Bandwidth and Colocation Provided by
I am working with a simple follow-me-style service: rather than have
something that rings several phones in turn, the user dials a number (in
the present implementation, unique to that user) to register his
presence at a particular extension.
What's the standard way to protect this from
On Mon, Sep 20, 2010 at 8:58 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com
wrote:
Anyone have a AudioCodes with Asterisk ???
I use many AudioCode devices with Asterisk. Mostly Mediant 1000s and
MP-114s,
I implemented a scenario where all extension have to pass trough a kind of
policy server (sql+ some script + dialplan) that enable/disable call
feature, taht is you can call XXX but you don't YYY.
I also added a Voicemail()-style + pin sceanrio to allow extensions to
access specific trunks.
Hope
On Monday 20 September 2010 11:01:36 Jose P. Espinal wrote:
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Hi all,
Can anyone help with the logic of which commands to use to say:
1. Extension is 600
2. See if has an ongoing call
3. Check if inbound or outbound to the extension
4. Find callerid of inbound call
Been reading http://www.voip-info.org/wiki/view/Asterisk+manager+API
Using latest 1.6.
You are right, the device is called Sipura SPA-3000. The settings are
factory-set, I haven't changed anything beside of SIP registration with the
asterisk.
How can I enable SIP debug?
--
Arie
On Mon, Sep 20, 2010 at 18:01, Paul Belanger
paul.belan...@polybeacon.comwrote:
On Mon, Sep 20,
I got confused while reading the documentation.
Tilghman Lesher wrote:
On Monday 20 September 2010 11:01:36 Jose P. Espinal wrote:
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized
That Apple App Store really gets imaginations going, doesn't it?
Wouldn't it be great to just publish an AGI script and see even 1%
of the Asterisk installed base buy it for the low, low price of $49?
Yes, that would be great. But one of the significant components of
the moneymaking App
On 16 September 2010 15:03, Jerry Geis ge...@pagestation.com wrote:
Jerry Geis wrote:
below is the results of the command.
grep -r ztconfig /etc/.
grep: /etc/./httpd/run/asterisk.ctl: No such device or address
grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address
What could we do to make the AsteriskExchange more effective?
Rod,
I'm not involved with Digium or even Asterisk on a daily basis so I
don't know you, I also don't know your intentions but taking you at face
value and answering your questions - I suggest you listen to the phone
call and
On 20/09/10 3:06 AM, Kevin P. Fleming wrote:
There is no fee to list free products on AsteriskExchange.
The main problem is the fee required to list non free products.
If the fee was a percentage of the sale price then I'm sure it would
work much better.
Otherwise it becomes a catch 22.
Hi all,
Sorry for the crosspost but I assume this may be of interest to both
businesses and users.
The Daily Asterisk News (running since 2004) is now accepting article
submissions.
Basically I've created a submission form where you specify whether your
post is commercial or non commercial
Thanks, Dean. I was able to listen to that conference live.
Digium's current licensing server has some limitations that make it
unsuitable for general use. We are investigating options to improve
the licensing platform, but have nothing to announce today. Even if
we did, it would be only one
Hi
Thanks for help.
I will try to help. But others might know more. What SIP client are you
using - a softphone, a hardphone? It looks like the client is sending
the full at 192.168.0.1 instead of just as the username.
Sebastian
That's right.hardphone is sending at
On Mon, Sep 20, 2010 at 10:41 PM, Rod Montgomery rmontgom...@digium.com wrote:
Does anyone reading this have an opinion on whether commercial
listings for complementary products and services should appear
directly on Asterisk.org?
Personally, I would like to see less commercial marketing on
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