Hi
I've been noticing an odd issue with our servers (1.4.17) where a large
number of one particular customer's (we operate a hosted VoIP platform)
calls go through a Local channel rather than the SIP channel and
whenever this happens our asterisk CDR is recording a billsec value of 0.
Our
Dear All,
I Have one test scenario where one of my kamailio servers is in Europe and
Another is in Singapore.
extensions can be registered to any of these 2 kamailio servers in the same
domain. now i want to send a call to an extension from Asterisk to one of
these kamailio servers based on
Well, how about piping an internet stream into a phone call via some app
in an extension?
Alex Balashov abalas...@evaristesys.com wrote:
cov...@ccs.covici.com wrote:
Is there any app to pipe a stream to a call either a meetme conference
or even a regular call?
Do you mean piping
It seems to me this could be achieved in the manner I just described
by specifying an Internet stream as a music on hold source and putting
the outgoing channel on hold via a Local channel.
--
Sent from mobile device
On Nov 16, 2009, at 5:36 AM, cov...@ccs.covici.com wrote:
Well, how about
I suppose that would depend on how the information about the
registrations is organised; do you want Asterisk to query some sort
of database used for backing these registrars and figure out where the
contact binding for a given AOR resides? AGI and func_odbc provide
fine ways to do that.
Hello,
I am installing dahdi on a machine
lspci
00:00.0 Host bridge: Intel Corporation 3200/3210 Chipset DRAM Controller (rev
01)
00:01.0 PCI bridge: Intel Corporation 3200/3210 Chipset Host-Primary PCI
Express Bridge (rev 01)
00:06.0 PCI bridge: Intel Corporation 3210 Chipset Host-Secondary
Has Asterisk any protection against brute force attack for SIP authentication?
Something like a maximum login attempt limit
Thanks
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To
sean darcy wrote:
Leif Madsen wrote:
sean darcy wrote:
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
Before I file a bug, is there anything I'm missing?
Does this happen on earlier versions of the 1.6.0 series prior to this
release
candidate?
I found that on a clean boot, I could not connect to Postgresql either.
In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the
module, and that seems to work. After bootup, cdr_pgsql.so is able to
connect immediately.
--
James Texter III
Sr. Software Engineer
NOBLE SYSTEMS
4151
James Texter wrote:
I found that on a clean boot, I could not connect to Postgresql either.
In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the
module, and that seems to work. After bootup, cdr_pgsql.so is able to
connect immediately.
This sounds as though you have
Hi all,
I have a problem with 1.6.1.7-rc1 and ENUM (with an own PowerDNS server
and NAPTR record). Maybe somebody has more experience with this or can
give me some input.
The dialplan:
exten = 292,1,Set(DIAL_NUMBER=43660123456)
exten = 292,2,Set(sip=
I thought that too, but I already checked. Postgresql is priority 64,
Asterisk is priority 90. Watching the boot sequence, I can see that
Postgresql is clearly started before Asterisk. It may be that there is
something in my config that causes this happen, but it's reproducable on
any box I
Most of the OS solutions I've seen require PHP and MYSQL. If you set up a
daemon/script to feed to mysql on a common server, most of this would work
fine. From my experience, I'd prefer some sort of central script
(C/Perl/PHP) that uses AMI to talk to the other boxes.
_
From:
Alex Balashov wrote:
As far as I know, Asterisk has no way to restrict the content of the
domain portion of the Contact URI. However, most commercial SBCs
should have a way to filter this, and it is highly recommended that
you do so.
It does actually; we added it to address this very
Hi,
We are using MixMonitor to record the call. When the call is bridged, the
latency is significant. We tried to increase the internet speed and the
server RAM and processor speed and still we are having that issue.
We use VoiceTrading and Gafachi's Termination minutes to make calls. As
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
thanks
---
Marek Cervenka
===
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- exten = s,x,Set(CDR(userfield) = information) - replace information
with the information like ${remoteip}
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
On Mon, Nov 16, 2009 at 9:40 AM, Bharath B. Reddy Bynagari
bynag...@mavensphere.com wrote:
We are using MixMonitor to record the call. When the call is bridged, the
latency is significant.
$ConversationFile =
$ConversationPath.conv_.$CallQID-$ConversationID.wav;
$self-agi-answer();
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a
hotdesk type system where anyone can log on to an extension - however what
I would love to do is relabel the phone with the current owner when this
I'm pretty sure it only pulls the background image during a reboot.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan
Thurman
Sent: Monday, November 16, 2009 9:20 AM
To: Asterisk Users Mailing List -
Shaun,
Thanks for your feedback. See my inline comments.
On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffell sruff...@digium.com wrote:
It appears there may be a regression in dahdi-linux 2.2.0 with regards to
the wcte12xp driver and the VPMADT032 module (as discussed
On Mon, Nov 16, 2009 at 7:29 AM, Peder pe...@networkoblivion.com wrote:
I'm pretty sure it only pulls the background image during a reboot.
On a 79x0, yes. On the 79x1 phones the user can change the background
to a list of custom images that you provide. It downloads the image
on the fly, and
If you just want to change the SIP configuration on the phone remotely there
are a few prerequisites:
1) Upgrade your phone's firmware to a recent release.
In the phone's config file on the tftp server:
2) Set telnet_level set to 2. Make sure it stays at 2 when you create your new
config file
- exten = s,x,Set(CDR(userfield) = information) - replace information
with the information like ${remoteip}
${remoteip} variable doesnt exist in asterisk (for remote voip phone)
SIPURI=sip:6...@192.168.1.184:5061 doesnt have public ip
i'm only found way
- check ${CHANNEL} for name
- check
snip
Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14
that uses your ability to press keys on the phone. You could apply
the same idea to press the correct buttons to change the background
without rebooting.
I can't find the script that I found to do this, but I'll keep
CLI output of calls that go through the local channel instead of the defined
channel would be useful to help diagnose what's going on here.
Thanks,
--Warren Selby
On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I've been noticing an odd issue with our servers
It is not the VLAN problem. Simply reboot of the 79xx takes up to 3 minutes and
we found no way to speed it up (phones works with Call Manager and VLAN's are
not implemented).
There are some other methods to display content on the phone screen without
editing local configs. Check
fail2ban
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
2009/11/16 Xavier Mesquida xavi...@yahoo.com
Has Asterisk any protection against brute force attack for SIP
authentication?
Something like a maximum login attempt limit
Thanks
snip
There are some other methods to display content on the phone screen without
editing local configs. Check http://www.ciptec.co.uk/ - commercial site but
shows the way.
/snip
If you just want to display user info on the phone, why not use the idle url
feature:
You're right - this doesn't sound like VLAN problem (but maybe a VLAN issue is
slowing the firmware download down, I can't tell. Can you?). This is a
problem of trying to reproduce extension mobility like features in
CallManager and the like.
You're right - rebooting does take a while. In
On Monday 16 November 2009 04:01:22 am Ishfaq Malik wrote:
I've been noticing an odd issue with our servers (1.4.17) where a large
number of one particular customer's (we operate a hosted VoIP platform)
calls go through a Local channel rather than the SIP channel and
whenever this happens our
Hello everybody,
I need help, I have a problem with conferences in asterisk, when many
people are in a conference sometimes there're users pressing phone keys
and this action emits a sound (DTMF of the phone keys), so, I need to
find the way of not listening this sound.. I'm using
Hello.
Sorry to repost this message but, I don't have the original message in my inbox
nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server
use a voip provider and a pstn. Pstn works fine but, I cant even connect to my
provider's server. I don't
On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez dlo...@palosanto.com wrote:
Hello everybody,
I need help, I have a problem with conferences in asterisk, when many
people are in a conference sometimes there're users pressing phone keys
and this action emits a sound (DTMF of the phone keys), so,
We have setup an * box for a small client with 10 phones. They have a
4500/500k ADSL connection which works great when no more than 8 external
calls are in progress. (ulaw)
The problem is when all 10 people try to use an external channel, AND/OR, 8+
incoming calls arrive at once. The symptom
Could it be your using option X when you have no extensions for the user to
exit to (therefore when they press dtmf instead of one and done, they just
keep going?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises
Hi All,
Currently I have voice calls from a certain SIP peer coming into an asterisk
server where the specific [SIP] channel is set to 'canreinvite=no'.
I would like to enable reinvites for certain calls, matched on DID. So I'm
wondering if there is a mechanism in the dial plan to turn on/off
Hello Everyone,
I'm looking for help/ideas on how to do the following:
I have a couple of people out of many (the couple of people randomly change)
who log into an on-call queue. A call comes in and it rings the on-call
extensions, but no one answers. I would like the call to then try the
It should be realistic, but have you considered just using followme to add
the cell phones to the queue list?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis
Elsberry
Sent: Monday, November 16, 2009 3:25 PM
To:
Do i rely need a pbx-card to use astersik for voicemail-system?
Or can i only use my internal pci-modem
A-link
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To UNSUBSCRIBE or update options
I had looked at followme as a solution but ran into the same stumbling block of
having to hard code the cell phone list. I didn't see a dynamic way of the list
being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on
Tuesday without editing the extensions.conf file
Since followme is extension-based, you have at least two options. Option
1 is to have a few extensions designated for following where you punch in
the cell numbers as you wish. Option 2 is to use day logic to point to
the following guys based on days.If I were doing option 2, I'd try to
use
Hi Travis,
There's lots of different ways to attack on-call roster solutions in Asterisk
- as Danny suggested, FollowMe() is definitely an option (and normally the
best), but it doesn't always suit the business need. However, also as Danny
suggested, in most cases using ASTDB in some way to
mattias wrote:
Do i rely need a pbx-card to use astersik for voicemail-system?
Or can i only use my internal pci-modem
If you want to connect to an analog phone line, then you will need to use some
sort of hardware to connect it (no, a PCI modem is not possible*) -- this can
either be a PCI
The full VoIP Security webinar from last Friday is now available on
Asterisk.org
http://www.asterisk.org/security/webinar
Thanks,
-S
Steven Sokol
Digium, Inc. | Marketing Director - Asterisk
1568 South Yorktown Place – Tulsa, OK – 74104
direct: +1 256-428-6101
mobile: +1
Hi Michael,
Your web interface for the on-call roster is pretty close to what we're
trying to trying to achieve. I would like to have people signing into the
on-call queue be the method that determined whose cell phone to call. I was
hoping there was a way to pass the call exiting the queue
I was previously using an old computer running Asterisk 1.2 with
zaptel. Once the CPU fried I switch to a new computer and I chose
AsteriskNow 1.5 running in 64bits to simplify the installation
process. I manage to find my way with configuring dahdi instead of
zaptel and to switch all my
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk. How would I do this?
The voipinfo wiki shows playing a congestion tone to the caller, but that
seems stupid since
Again – lots of ways to do it – you could use a web interface to set the
numbers in ASTDB for lookup – or you could create an IVR to ask for the number,
and store it in ASTDB that way.
Good luck!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.comwrote:
snip
I don't know what else to try. When I try to call I get this at the cli:
== Using SIP RTP CoS mark 5
-- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40,
SIP/1xxx763x...@voipprovider) in new stack
==
How can I set a maximum call duration on a ZAP channel?
Thank you.
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Hi all,
does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two
different accounts on the same server (i.e. two different extensions)?
I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something.
The phone sends SIP packets from a high-numbered UDP port but expects
a reply on
You need to enable SIP transformations on the firewall, the packets will
have to be dynamically re-written to handle multiple Cisco phones of
these models. Be sure 'nat=no' is set in sip.conf for the phones as
well, or Asterisk will reply to the incorrect ports (source instead of
the mangled
Darryl,
OK, that could work but it makes the use of these phones behind
consumer routers rather impossible. How many of those will inspect and
transform SIP packets? Oh why does Cisco have to do things differently
from everyone else...
Luki
2009/11/16 Darryl Dunkin ddun...@netos.net:
You need
On Mon, Nov 16, 2009 at 10:53 PM, Luki lugos...@gmail.com wrote:
Darryl,
OK, that could work but it makes the use of these phones behind
consumer routers rather impossible. How many of those will inspect and
transform SIP packets? Oh why does Cisco have to do things differently
from
But are not pbx card and modem the same?
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Leif Madsen
Skickat: den 16 november 2009 23:46
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re:
2009/11/17 Martin Roy m...@mac.com
I was previously using an old computer running Asterisk 1.2 with
zaptel. Once the CPU fried I switch to a new computer and I chose
AsteriskNow 1.5 running in 64bits to simplify the installation
process. I manage to find my way with configuring dahdi instead
2009/11/17 Michelle Dupuis supp...@ocg.ca
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk.
On which tech, does this trunk rely ?
Is it a SIP trunk ?
How
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