Hello all,
I know it was on the list before but i have some questions about the
Kirk IP600v3, the requested configuration files were send private i guess
Does anybody have the correct SIP settings for handsets connected to the
Kirk. IP600v3
I am particulair intrested in settings regarding:
Hi,
this problem could be a bit tricky. We´ve got some good old Fax-Machines
here and need to create an fax-report of all faxes which goes out through
this devices including an copy oft he fax. The fax-machines are not able to
store this fax-report on an storage-place ore send it to an
We're doing callback here. Asterisk dials a number, waits for an answer,
plays a prompt, dials a second number, and bridges the channels together.
Calls are initiated from the AMI.
No problems there. Easy stuff.
We want to generate two accounting records for the bridged calls so that the
user is
I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue or
isdn ) since it's always the same way of configuring
- define ports
- define interface
- define services
my config:
1 asterisk
1 patton ( let say 4638 )
the patton gw registers itself has a asterisk sip peer,
inside the
2008/1/10, Benny Amorsen [EMAIL PROTECTED]:
Steve Langstaff [EMAIL PROTECTED] writes:
I agree that sending an OPTION message from the Asterisk server could
well have a low processing load.
The original poster wanted to use OPTIONS sent FROM the Asterisk server
to query the phone
2008/1/10, Benny Amorsen [EMAIL PROTECTED]:
Olivier [EMAIL PROTECTED] writes:
To get a polite go to hell ! in return ? ;-)
I think the vendors will be nicer than that.
You're right.
Asterisk has a good bit
of the VoIP PBX market.
Asking all of them for guidance (how do you plan to
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue or
isdn ) since it's always the same way of configuring
- define ports
- define interface
- define services
my config:
1 asterisk
1 patton ( let say 4638 )
the patton
Mitel and Avaya support 802.1X with proprietary protocols.
For Siemens, I'm not so sure.
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2008/1/10, Robert Moskowitz [EMAIL PROTECTED]:
Jeronimo Romero wrote:
Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
This is so unlikely. I worked on 802.1X and 802.11i. There is just too
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are implemented in DECT base stations) ?
Regards
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On 08:35, Thu 10 Jan 08, IT-Connect wrote:
I've run Asterisk 1.4.17 with mISDN 1.7 on a Suse 10.0 with
Kernel-Version 2.6.23.13. But there are any issues
with newer Kernel-Versions. You have to patch the mISDN packet.
If you're interested, you can get a description from me.
Debian etch by
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 10, 2008 6:44 AM
Subject: Re: [asterisk-users] OT: Traffic Shaping
-BEGIN PGP SIGNED MESSAGE-
Hash:
On 11:22, Thu 10 Jan 08, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are implemented in DECT base stations) ?
Roaming/handover functionality is
On 12:28, Thu 10 Jan 08, Robert Lister wrote:
On Thu, Jan 10, 2008 at 11:22:29AM +0100, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are
Olivier wrote:
Mitel and Avaya support 802.1X with proprietary protocols.
For Siemens, I'm not so sure.
Two facts:
Proprietary EAP methods that can actually complete in a reasonable
amount of time. Many of these have small security holes and thus are
not acceptable as standards. (I know, I
exactly
isdn patton - eth/lan sip asterisk
jl
2008/1/10, Olivier [EMAIL PROTECTED]:
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue
or isdn ) since it's always the same way of configuring
- define ports
- define
On Thu, Jan 10, 2008 at 11:22:29AM +0100, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are implemented in DECT base stations) ?
Yes. It is a capability
Am Donnerstag, den 10.01.2008, 12:31 +0100 schrieb Michiel van Baak:
On 11:22, Thu 10 Jan 08, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are
Olivier wrote:
I thought that :
1. 802.1X was mainly when you plug your hardphone into your network,
802.1X-2001 was written to secure ports on a 802.3 switch. Originally
for PCs works just fine for phones. Really does NOT play with VLANs,
but HP cheated (I know their lead engineers).
Hi to all
I'm a new user of TDM400P card. The configuration is OK and I have no problem
for incoming call. When I try to place a outgoing call towards a PSTN number
the call is not always answered. In other words TDM400P seems to not understand
that the call has been accepted from the remote
I have been trying to do this. The only thing, really holding me up is
IAXmodem 1.0 rpm for Centos5. Anyone want to build it for me (I am
terrible at compiling; do it maybe once a year).
Armin Krämer wrote:
Hi,
this problem could be a bit tricky. We´ve got some good old
Fax-Machines
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
exactly
isdn patton - eth/lan sip asterisk
so why is misdn installed for ?
it seems you don't have any ISDN card inside you Asterisk server.
jl
2008/1/10, Olivier [EMAIL PROTECTED]:
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]:
Olivier wrote:
I thought that :
1. 802.1X was mainly when you plug your hardphone into your network,
802.1X-2001 was written to secure ports on a 802.3 switch. Originally
for PCs works just fine for phones. Really does NOT play with VLANs,
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]:
Jeronimo Romero wrote:
Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
This is so unlikely. I worked on 802.1X and 802.11i. There is just too
2008/1/10, Michiel van Baak [EMAIL PROTECTED]:
On 11:22, Thu 10 Jan 08, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset
(then,
these functions are implemented in DECT base
2008/1/10, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Donnerstag, den 10.01.2008, 12:31 +0100 schrieb Michiel van Baak:
On 11:22, Thu 10 Jan 08, Olivier wrote:
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY
support
roaming and handover and are these
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
... I get the wierd impression that either both modules somehow
get interrupts from the two cards, or each module handles a
different card. This hsouldn't happen.
So try blacklisting one of them:
I've already done something like that: removing the
On Thu, Jan 10, 2008 at 02:31:53PM +0100, Jaap Winius wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
... I get the wierd impression that either both modules somehow
get interrupts from the two cards, or each module handles a
different card. This hsouldn't happen.
So try blacklisting
I am seeing slight differences in URIs.
In the case where things work, the URI is [EMAIL PROTECTED] where it
does not work is [EMAIL PROTECTED]:5060
In the first case I suspect that Asterisk did something, perhaps at
startup, where it 'decided' it was behind a firewall, so let the
firewall
Dovid B wrote:
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Erik Anderson wrote:
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I was thinking of doing. What's the traffic
On Thu, Jan 10, 2008 at 01:47:39PM +0100, Michiel van Baak wrote:
We have 2 different setups in production.
NEC-Philips ip-dect and the kirk/tiptel ip-dect.
The NEC-Philips one works with a dedicated server to
controll the registration etc, and all the radios are
connected to the normal
Olivier wrote:
When we were starting on 802.1AE (LinkSec), Norm Finn (a CISCO Fellow
and long time worker on 802.1 and other layer 2 standards) said it
well:
Layer 2 security protects and addresses the liablities of the
network owner
Layer 3 security protects and
Olivier wrote:
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Jeronimo Romero wrote:
Does anyone know if sip phones from any of the major IP phone
vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
On Thu, Jan 10, 2008 at 02:40:21PM +0100, Olivier wrote:
that is what I call handover
roaming = without any ongoing call
handover = with ongoing call
this would need the appropriate logic in the base
stations. I know such hardware exists (Kirk!?),
Kirk base stations support roaming and
On 14:15, Thu 10 Jan 08, Robert Lister wrote:
On Thu, Jan 10, 2008 at 01:47:39PM +0100, Michiel van Baak wrote:
We have 2 different setups in production.
NEC-Philips ip-dect and the kirk/tiptel ip-dect.
The NEC-Philips one works with a dedicated server to
controll the registration etc,
On Jan 10, 2008 8:24 AM, Drew Gibson [EMAIL PROTECTED] wrote:
It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN
ports, each can be assigned to a VLAN of your choosing and you can use them
as you please (at least you can under openWRT).
Yup - you can do the same with
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]:
Olivier wrote:
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Jeronimo Romero wrote:
Does anyone know if sip phones from any of the major IP phone
vendors
support 802.1x
I have one for centos 4 that I compiled and should work just fine on 5. It is
located here. http://www.taylortelephone.com/asterisk If you have problems
let me know and I will build one for you.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
ok sorry for the confusion created,
I mean isdn network , in other word tdm,
so tdm link connected to patton, patton connected in the lan via ethernet
speaking sip,
jl
2008/1/10, Olivier [EMAIL PROTECTED]:
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
exactly
isdn patton - eth/lan sip
Olivier wrote:
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Olivier wrote:
2008/1/10, Robert Moskowitz [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Jeronimo Romero
Jonn R Taylor wrote:
I have one for centos 4 that I compiled and should work just fine on 5. It is
located here. http://www.taylortelephone.com/asterisk If you have problems
let me know and I will build one for you.
I don't see IAXmodem 1.0 there...
Hi,
I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always
rings but sometimes (it happens randomly!) no voice is passing thru (2
ways).
Asterisk CLI shows this warning:
Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on
authentication for INVITE to
On Jan 9, 2008 11:44 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Erik Anderson wrote:
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I was thinking of doing. What's the traffic
shaping like? Can I specify
I just built one. Give it a try.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Moskowitz
Sent: Thursday, January 10, 2008 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using Asterisk as an
So, I think in this case (Ethernet link), standard spandsp doesn't help as
it needs a TDM board.
But, as spandsp has recently gained T.38 support, it could help to build
email2fax or fax2email but I have no experience myself in it.
I would be very curious to know.
Regards
2008/1/10, Jean-Louis
How would I go about doing this?
The way I have paging set up (just to test it), is:
[intercom] ; Paging context
exten = s,1,Answer
exten = s,2,Playback(beep)
exten = s,3,Set(TIMEOUT(digit)=5)
exten = s,4,WaitExten(10)
exten = *,1,SIPAddHeader(Alert-Info: Ring Answer)
exten =
Olivier wrote:
So, I think in this case (Ethernet link), standard spandsp doesn't
help as it needs a TDM board.
Nope not the case at all. I have been doing
fax--ATA--lan--Asterisk-email for quite some time without ANY zaptel
interfaces. Zaptel creats the pseudo interface and that does the
The only problem with this workaround is that on the Polycom 550 (backlit
display) the backlit goes bright every 30 seconds then back to dim. Any
work around for that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson
Sent: Friday, December
Jonn R Taylor wrote:
I just built one. Give it a try.
I will grab it and put it into my local repo after I get back from an
errand. thanks!
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert
Moskowitz
Sent: Thursday, January 10, 2008
What zap driver are you using? ztdummy?
PaulH
On Wed, 2008-01-09 at 15:41 -0500, Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm getting
fuzzy voice between the Polycom IP SoundStation 550 and the Asterisk
server. I've checked all of my codec settings
-BEGIN PGP SIGNED MESSAGE-
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Kristian Kielhofner wrote:
On Jan 9, 2008 11:44 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Erik Anderson wrote:
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I
Hello all,
I know this has been discussed before but I am not finding the thread on
voip-info or site:lists.digium.com through google.
I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on
openSuSE 10.2, Dual core AMD Opteron, purely SIP.
I haver been experiencing a problem
Doug,
That bug ID was a dead ringer. The workarounds in the bug worked
perfectly. BTW I'm on a openSuSE 10.3 system with gcc 2.4.1.
Thanks for the pointer.
Mike
On Jan 9, 2008, at 8:30 PM, Doug Lytle wrote:
Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm
At 14:54 1/10/2008, John Millican wrote:
Hello all,
I know this has been discussed before but I am not finding the thread on
voip-info or site:lists.digium.com through google.
I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on
openSuSE 10.2, Dual core AMD Opteron,
Nailed it!
TCPdump on Trixbox 2.4 (Asterisk 1.4.17-1) going out and public side of
firewall (Linksys WRT54G running Sveasoft) Firewall is configued NOT to
NAT (public addressing on internal network.
I stop asterisk (amportal stop). wait 30 min to insure timeout. Start
both tcpdumps. Start
How does one get asterisk to timeout realtime request via res_odbc to
unixODBC? I've set timeouts as appropriate for freetds (which
unixODBC is using.) However, it doesn't seem to work. It takes over 3
minutes to timeout a connection and queries never seem to timeout, so
a channel waiting
I need to setup multiple fax extension numbers.
What is the best way to do that?
It should send the fax as pdf to the assigned email address (or
addresses) of that extension number.
It should also move the fax to a web site for online view.
It should - if possible - try to make OCR text file as
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What do you mean by In Use?
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In
On Fri, Jan 11, 2008 at 12:45:48AM +0100, Jaap Winius wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What do you mean by In Use?
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
Hi Jaap,
On Thu, 2008-01-10 at 04:23 +0100, Jaap Winius wrote:
Hi list,
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
I've tried to get it to work on a
Ronald Wiplinger wrote:
I need to setup multiple fax extension numbers.
What is the best way to do that?
HylaFAX+ running on the Asterisk server
Fax machines and inbound DID attached to the [fax] context.
MySQL database lookup against extensions to pull email address.
Passes that
Hi all,
I just wanted to share my experience. I tried to mix up the Sangoma A200
with digium HPEC and the result has no changed like before I got the
HPPEC, even I had tried to complied the Zaptel code with the tuning that
is available.
For those who intend to do so, please dont. Btw, Digium
Hi everyone,
having a issue with asterisk and my new Voip providers service.
Iv set up many asterisk systems before but never seen this and have
tried to fix this with no luck..
I have used this exact same sort of setup for 5 other providers and
never had this issue, If i replace the trunk
Hi, i am having a problem with my point to point t1, which is being
resolved and is a seperate issue. sangoma support has been a huge help
and i am waiting on verizon to increase the signal output of the
smartjack.
But my issue is that in the meantime my fallover extensions arent
working. Well
Are they expecting numbers in a 61 format?
PaulH
On Fri, 2008-01-11 at 16:27 +1100, Kev S wrote:
Hi everyone,
having a issue with asterisk and my new Voip providers service.
Iv set up many asterisk systems before but never seen this and have
tried to fix this with no luck..
I
10 jan 2008 kl. 15.24 skrev Robert Moskowitz:
I am seeing slight differences in URIs.
In the case where things work, the URI is [EMAIL PROTECTED] where it
does not work is [EMAIL PROTECTED]:5060
In the first case I suspect that Asterisk did something, perhaps at
startup, where it
10 jan 2008 kl. 16.48 skrev gincantalupo:
Hi,
I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom
always
rings but sometimes (it happens randomly!) no voice is passing thru (2
ways).
Asterisk CLI shows this warning:
Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden -
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