On 07/22/2010 09:33 PM, Danny Nicholas wrote:
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Albert
> Scholtalbers
> *Sent:* Thursday, Jul
On 07/23/2010 01:12 AM, Karl Fife wrote:
>> Cassius Smith wrote:
>>> Hello all,
>>> I'm in final testing stages and preparing training for a new Asterisk
>>> rollout. I'm replacing a Cisco Call Manager system, and re-flashing
>>> the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call
On 07/22/2010 09:14 PM, Ken D'Ambrosio wrote:
> Hey, all. I'm looking -- if possible -- for a decent, multi-platform
> soft-phone. Specifically, Linux and Windows; that way, I'll go through
> the same issues my end users do. I've noticed a couple (e.g., minisip,
> which seems abandoned, and sip-
Hi Everyone,
Using a PRI with Sangoma A101D and Asterisk 1.4.2.x.
I notice that occasionally after a call is disconnected and both the phone
devices and the the channel is down but the bridge stays open for hours.
Channel Location State Application(Data)
Local/9054445.
On Thu, Jul 22, 2010 at 2:46 PM, Matt wrote:
> I've got an interesting situation where I have one cable run from the feed
> area to the service area. I have three devices that I need to power at the
> service area. Is anyone aware of a device that will take the POE from the
> cable run and then
Sometimes it doesn't matter what the device actually draws as much as
what it declares itself to be to the upstream switch. For example, most
polycom phones draw under 9 watts but in practice they declare
themselves to the switch as requiring the full 15.4 watts allowed by
802.11af class 1.
I've s
On 21:46 Thu 22 Jul , Steve Underwood wrote:
> It might help if you explained what you expect those pages should look
> like. I see three quite plausible pages.
I expect to see this http://imagebin.ca/img/Eihpy0.jpg
--
_
--
Hi,
Zoiper is a great software to have both SIP and IAX. As a beginner to
Asterisk, I find very well but to my understanding it does not have linux
version.
X-lite have both Windows and Linux but it is a bit clumsy to set up.
CK
On Fri, Jul 23, 2010 at 5:04 AM, Ronaldo Zacarias Afonso <
ronaldo
AMARDEEP SINGH wrote:
> Hello All,
>
> Scenario:
> -We use asterisk as voicemail server for our cellular network.
> Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
> -Extensions in * are virtual, just for leaving and accessing voicemail.
>
> Requirement:
> Asterisk to sen
> Cassius Smith wrote:
>> Hello all,
>> I'm in final testing stages and preparing training for a new Asterisk
>> rollout. I'm replacing a Cisco Call Manager system, and re-flashing
>> the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call,
>> I use the "Confrn" softkey to invite othe
The Aastra 53i draws only 2 Watts from a Linksys 24 port POE switch. 25
phones is around 55 Watts.
-Bruce
On Thu, Jul 22, 2010 at 5:16 PM, Andrew Latham wrote:
> The Snom 360 phone in front of me draws 4w...
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
> * Learn more about OSS http:/
- Original Message -
From: "David Gibbons"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, July 22, 2010 1:58 PM
Subject: Re: [asterisk-users] POE Splitters
> There is no such device -- it's outside of the POE spec.
>
> Class 3 devices are allowed to con
Hello All,
Scenario:
-We use asterisk as voicemail server for our cellular network. Asterisk box
is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
-Extensions in * are virtual, just for leaving and accessing voicemail.
Requirement:
Asterisk to send SMS to cell switch(running SMSC) on
Cassius Smith wrote:
> Hello all,
> I'm in final testing stages and preparing training for a new Asterisk
> rollout. I'm replacing a Cisco Call Manager system, and re-flashing
> the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call,
> I use the "Confrn" softkey to invite other pa
The Snom 360 phone in front of me draws 4w...
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Jul 22
Hello all,
I'm in final testing stages and preparing training for a new Asterisk
rollout. I'm replacing a Cisco Call Manager system, and re-flashing
the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call,
I use the "Confrn" softkey to invite other participants. I can add one
Hi Ken,
Can it be an IAX client?
If so, I'd recommend KIAX. I used it once, both on Linux and Windows,
and it worked for me.
[]s
Ronaldo.
On Thu, Jul 22, 2010 at 4:14 PM, Ken D'Ambrosio wrote:
> Hey, all. I'm looking -- if possible -- for a decent, multi-platform
> soft-phone. Specifically, L
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Thursday, July 22, 2010 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Soft phones
Zoiper seems to have a software update every other week, and annoys you to
death on updates, and sometime the update breaks it.
I am looking myself for a good windows softphone, Zoiper is nice, never
tried the pay for version.
-Original Message-
From: asterisk-users-boun...@lists.digium
vijay kumar wrote:
>
> How are you doing these days!
I just love the clueless.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
--
How are you doing these days!
I bought some Bag and Watches on one great online shop www-asomart-com
They provide all kinds of goods and many brands.
2010 Summer Shopping Guide (www-asomart-com )
Jessica
2010723352--
___
On Thu, 22 Jul 2010, Ken D'Ambrosio wrote:
> Hey, all. I'm looking -- if possible -- for a decent, multi-platform
> soft-phone. Specifically, Linux and Windows; that way, I'll go through
> the same issues my end users do. I've noticed a couple (e.g., minisip,
> which seems abandoned, and sip-co
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Scholtalbers
Sent: Thursday, July 22, 2010 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [AsteriskNow] Errors with
Hi Danny,
I think the enemy comes from the inside ;)
Well as long is it is *your* firewall it shouldn't be to difficult to change
the settings.
Otherwise post your email address and I can mail you the 440Kbytes
personally
> My firewall is set up to reject non-us sites. Sorry.
>
>
--
_
Hello list,
For a customer I need to detect blue alarms on his T1 trunks. His server is
in running asterisk and zaptel 1.4. I have a tool to generate all sorts of
alarms, but on generating blue alarm, zaptel recognizes them as red alarms.
This is not good for us as we need to get blue alarm as blu
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Scholtalbers
Sent: Thursday, July 22, 2010 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [AsteriskNow] Errors with
Hey, all. I'm looking -- if possible -- for a decent, multi-platform
soft-phone. Specifically, Linux and Windows; that way, I'll go through
the same issues my end users do. I've noticed a couple (e.g., minisip,
which seems abandoned, and sip-communicator, which, honestly, is probably
a great IM
Hi Danny,
What does no dice mean in this case. I know they use them in Vegas. ;)
The site is non US, it is the Netherlands. Do you have problems downloading?
> Ø No dice – is 92.254.55.200 a non-US site?
>
--
_
-- Band
Will your single POE port even supply enough power to three devices
without complaining?
Plus, what about the data?
Sounds as if you will need a smaller switch in the service area to
supply power and data
John Novack
Matt wrote:
> I've got an interesting situation where I have one cable run f
There is no such device -- it's outside of the POE spec.
Class 3 devices are allowed to consume at max 15.4W. Most phones are class 3
devices. The math just doesn't work out. Even if you used the draft standard
for class 4 (~30W), you could still power max 2 devices at 15W/ea.
-Dave
On Thu, Jul
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Scholtalbers
Sent: Thursday, July 22, 2010 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [AsteriskNow] Errors with
> Can you post the op_server.pl somewhere? I don’t have the resources to do
a full ANOW install to look at it.
http://92.254.55.200/op_server.pl
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New to A
I've got an interesting situation where I have one cable run from the feed
area to the service area. I have three devices that I need to power at the
service area. Is anyone aware of a device that will take the POE from the
cable run and then allow me to split it to two or three devices at the
s
Can you post the op_server.pl somewhere? I don't have the resources to do a
full ANOW install to look at it.
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introduc
Hi Danny,
This is what I tried to do, printing the values of the variables.
1) This didn't work with print (probably it only prints STDERR)
2) I tried to print it to a file in /tmp dir (no result)
3) Than I tried to change that part of the file and noticed that it did not
have any influence.
4) So
>We know it has to do with the graphical panel interface of Asterisk and
don't influence the performance of Asterisk. Even moving the line 3367 in
/var/www/html/panel/op_server.pl a few lines down don't alter the warnings
(after restarting asterisk and rebooting the server)
>After did we up
Hi there,
We did a clean install the AsteriskNOW 1.7.0 64 bits ISO and configured it.
On the main screen (Crtl-ALT-F1) we keep seeing the following lines when
making a call
Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in con
We dont have any Digium cards, we just have a GrandStream FXS 8-port device
with 2 analog phones and one Grand stream FXO 8-port device with one POTS
line and both are connected to the netgear switchvery rarely the analog
phones are used and its very rare that calls are made through POTS using
On Jul 21, 2010, at 7:05 PM, Apu Islam wrote:
> Can any good men on this group share me the firmware of a Cisco 7960 Phone?
> Currently this one has Call Manager Firmware installed, I am trying to
> convert it into SIP.
> Much appreciated.
>
>
> Apu
Try google keywords: "index of" P0S3-06-3-
Hi list,
I am looking to set up IVR for company in NM. Any good and
especially stay away from recommendations for providers?
Thanks
Danny Nicholas
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Hi,
I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ
as target user which is registered.
Asterisk server IP: 70.118.x.x
calling user IP: 117.58.x.x
called user IP:117.58.x.x:5062
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,3
Christian wrote:
> Hi all,
> Does anyone know any good SIP based provider that offers free calling
> within europe for some monthly fee?
> Many thanks!
>
It will always depend on call volume and a fixed monthly fee may not
always be the best value. A lot of ITSPs have a monthly charge almost
t
Paul Belanger wrote:
> On Thu, Jul 22, 2010 at 6:53 AM, Chandrakant Solanki
> wrote:
>> Is anything missing in above configuration or something goes wrong.?
>>
> kamailio != asterisk
>
> Wrong list.
>
Looks like opensips from the code that was pasted.
--
__
On 07/22/2010 12:15 PM, Alexander Aksarin wrote:
> On 09:06 Thu 22 Jul , Alexander Aksarin wrote:
>
>> Hello to all. I have succesfully received fax by app_fax, but tif files are
>> weird.
>> There a faxes sended by several fax machines to asterisk.
>> http://filebin.ca/hnnumf/122.tif
>> h
Zhang Shukun wrote:
> hi,list
> Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after
> i make and make install. i cant find the .so file.
>
> is this mean it can't install on 64bit Cent-OS. ps: it works fine on
> the 32 bit Cent-OS
>
> Thanks very much!
>
I have a live system r
On Thu, Jul 22, 2010 at 6:53 AM, Chandrakant Solanki
wrote:
> Is anything missing in above configuration or something goes wrong.?
>
kamailio != asterisk
Wrong list.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon
On Thu, Jul 22, 2010 at 2:25 AM, Zhang Shukun wrote:
> Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after
> i make and make install. i cant find the .so file.
>
Yes, I believe they are installed to /usr/lib/asterisk/modules under
CentOS. What does the output of make install show
hello
I debuged asterisk RTP packet,it seems to delay.
Anyone has any ideas on this matter?
localhost*CLI> Got RTP packet fromxx.xx.3.224:3456 (type 00, seq 037872,
ts 1336569504, len 000160)
Sent RTP packet to xx.xx.3.224:3456 (type 00, seq 026327, ts 137432, len
000160)
localho
Hello,
looks like sipvicous. there is allready a new version to break such
attacks using sipvicous.
http://blog.sipvicious.org/
best regards.
steve smith
mosbah abdelkader schrieb:
> An attacker is scanning my Asterisk Switch to gain illegitimate access
> to VoIP call functionality.
>
>
> Us
Hello
I need *count* for number of active calls on kamailio server. I have done
following configuration in my kamailio.cfg file
...
loadmodule "dialog.so"
modparam("dialog","profiles_with_value","caller")
modparam("dialog", "dlg_flag", 4)
route[0] {
...
if(is_method("INVIT
On Thu, 22 Jul 2010, mosbah abdelkader wrote:
> An attacker is scanning my Asterisk Switch to gain illegitimate access to
> VoIP call functionality.
> Please help me resolve this problem.
Read The Fine Archives.
And more importantly, if you have not updated your sip.conf file to add
in:
al
Have a look at fail2ban
mosbah abdelkader wrote:
> An attacker is scanning my Asterisk Switch to gain illegitimate access
> to VoIP call functionality.
>
>
> Using a sip scanning tool, *it* sends REGISTERs with random identities.
> And when it discovers one identity subscribed in my switch, it
An attacker is scanning my Asterisk Switch to gain illegitimate access to
VoIP call functionality.
Using a sip scanning tool, *it* sends REGISTERs with random identities. And
when it discovers one identity subscribed in my switch, it tries to
authenticate with random passwords using this user nam
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Steve Edwards
> Sent: Wednesday, July 21, 2010 9:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Ci
I would appreciate it if you didn't top-post.
das sandesh writes:
> Hi Benny...
>
> DTMF tones are heard at the SIP phones side and not the other
> party...'server side' means from the Asterisk side.from the
> wireshark captures it appeards that the dtmf digits were sent from the
> aster
On Wed, 21 Jul 2010, Leif Madsen wrote:
> On 10-07-16 02:38 PM, Anita Hall wrote:
>> Is it possible to receive video calls using Asterisk and then process
>> them as an IVR ? One of our clients wants to set-up a video IVR system
>> in the US and we are evaluation possible options.
>>
>> Also, what
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