On Fri, Jan 9, 2009 at 4:46 PM, Jeff LaCoursiere wrote:
>
>
> On Fri, 9 Jan 2009, James Noble wrote:
>
> >
> > I had the same problem with a sangoma card and a clean install of
> asterisk
> > as well as a trixbox set up. I finally started using a vegastream to
> handle
> > the T1 connections and
Dave Platt wrote:
>> I may be over simplifying but I would have a serial number object that
>> gets incremented anytime it is called and will be set to 0 at start-up.
>> I would then use it to generate a UUID like this:
>> MAC.serialid.64bit timedate
>>
>
> I suggest reviewing RFC 4122, whic
Tilghman Lesher wrote:
> On Friday 09 January 2009 13:52:56 Anthony Francis wrote:
>
>> Tilghman Lesher wrote:
>>
>>> We are entirely interested in DETERMINISTIC methods of uniqueness, not
>>> random and hope-for-the-best. Given a truly random generator, it is
>>> possible for the same num
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instru
Vieri wrote:
> --- On Fri, 1/9/09, Tilghman Lesher wrote:
>>> --- On Fri, 1/9/09, Josiah Bryan
>> wrote:
Is this right after bringing online the alias IP?
If so, you might try using arp-sk to broadcast an
>> ARP
packet to kick-start the IP lookup...
http://sid.rstack.org/arp-s
Hi,
While issuing make, I've got:
...
CC [M] /usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.o
/usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.c: In function âzt_registerâ:
/usr/src/zaptel-1.4.11~dfsg/kernel/zaptel-base.c:5230: error: implicit
declaration of function âclass_device_createâ
My sys
On Fri, Jan 9, 2009 at 7:34 PM, Jeff LaCoursiere wrote:
>
>
> On Fri, 9 Jan 2009, Steve Totaro wrote:
>
>>
>> $5k for a single T1 is/was pretty much the norm. Go price non-used T1
>> cards for big proprietary phone systems.
>>
>
> Thats a copout. Big proprietary phone systems are expensive by de
On Sat, 10 Jan 2009, Tzafrir Cohen wrote:
>>
>> Its not a PRI. Its an RBS T1 with E&M Wink. I will try enabling the SIP
>> debug, though, that is a good idea. Is there any kind of extra debugging
>> for RBS T1?
>
> No idea, but the driver is much more aware of the specifics. So maybe
> their
On Fri, 9 Jan 2009, Steve Totaro wrote:
>
> $5k for a single T1 is/was pretty much the norm. Go price non-used T1
> cards for big proprietary phone systems.
>
Thats a copout. Big proprietary phone systems are expensive by default -
certainly not to be considered "the norm".
I say it is an e
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Hash: SHA1
Tilghman Lesher wrote:
> Mark erroneously assumed there would be another release candidate, which
> there was not. So while it's not in 1.6.0.3, it will be in the 1.6.0.4
> release, when that occurs.
Thanks Tilghman. I wait with bated breath.
Best
On Fri, Jan 9, 2009 at 6:46 PM, Jeff LaCoursiere wrote:
>
>
> On Fri, 9 Jan 2009, James Noble wrote:
>
>>
>> I had the same problem with a sangoma card and a clean install of asterisk
>> as well as a trixbox set up. I finally started using a vegastream to handle
>> the T1 connections and was able
On Fri, Jan 09, 2009 at 10:33:44PM +, Jeff LaCoursiere wrote:
>
>
> On Fri, 9 Jan 2009, Andres wrote:
>
> [snip]
>
> >> I have the "full" logging enabled, and here is an excerpt of a call that
> >> was terminated. You can see the conversation lasted about forty seconds
> >> before it was hu
On Fri, 9 Jan 2009, James Noble wrote:
>
> I had the same problem with a sangoma card and a clean install of asterisk
> as well as a trixbox set up. I finally started using a vegastream to handle
> the T1 connections and was able to get rid of the problem.
>
> James
>
$5K for a sinlge T1? Tha
On Fri, Jan 9, 2009 at 2:33 PM, Jeff LaCoursiere wrote:
>
> [also posted on Trixbox trunk forum]
>
> I am also working with Sangoma directly to debug this, but so far no real
> luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE
> 3.2.6 (3.2.7 is out, but nothing has changed th
> I may be over simplifying but I would have a serial number object that
> gets incremented anytime it is called and will be set to 0 at start-up.
> I would then use it to generate a UUID like this:
> MAC.serialid.64bit timedate
I suggest reviewing RFC 4122, which discusses UUID formats in some
--- On Fri, 1/9/09, Tilghman Lesher wrote:
> > --- On Fri, 1/9/09, Josiah Bryan
> wrote:
> > > Is this right after bringing online the alias IP?
> > > If so, you might try using arp-sk to broadcast an
> ARP
> > > packet to kick-start the IP lookup...
> > > http://sid.rstack.org/arp-sk/
> >
> > T
On Friday 09 January 2009 14:56:17 Barry L. Kline wrote:
> Mark Michelson wrote:
> > Thanks for pointing this out. I have located the erroneous code and have
> > fixed it in subversion, revision 161490. The next rc of 1.6.0 will not
> > have this bug.
>
> This bug still exists in the recently-relea
On Fri, 9 Jan 2009, Steve Totaro wrote:
> It looks normal to me. I think two dropped calls a day is reasonable
> and I would start looking for commonalities.
I tried that logic - they don't buy it :) The sad part is I replace a
Nortel system that did NOT have the issue (according to their re
On Fri, 9 Jan 2009, Andres wrote:
[snip]
>> I have the "full" logging enabled, and here is an excerpt of a call that
>> was terminated. You can see the conversation lasted about forty seconds
>> before it was hungup.
>>
>>
> What you need to do is figure out who is ordering the call to be
> han
Jeff LaCoursiere wrote:
>[also posted on Trixbox trunk forum]
>
>I am also working with Sangoma directly to debug this, but so far no real
>luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE
>3.2.6 (3.2.7 is out, but nothing has changed that would affect this
>problem). The
On Fri, Jan 9, 2009 at 4:33 PM, Jeff LaCoursiere wrote:
>
> [also posted on Trixbox trunk forum]
>
> I am also working with Sangoma directly to debug this, but so far no real
> luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE
> 3.2.6 (3.2.7 is out, but nothing has changed tha
On Fri, Jan 09, 2009 at 04:05:01PM -0500, John Todd wrote:
>
>
> Dilemma: Digium will sometimes receive requests to send GPG-encrypted
> mail dealing with security issues. This works somewhat poorly for
> email role accounts where there are multiple recipients on a single
> address. If th
[also posted on Trixbox trunk forum]
I am also working with Sangoma directly to debug this, but so far no real
luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE
3.2.6 (3.2.7 is out, but nothing has changed that would affect this
problem). The system gets about 200 calls in
On Fri, Jan 9, 2009 at 3:57 PM, wassim Darwish wrote:
> hi:
> When iam sending calls through sip a fake ringback tone is generated and
> then call status can't be viewed (if call is ringing,busy,offline) it just
> rings and rings.
> Can i disable this?
>
> Thanks in advance.
>
If you are using th
Dilemma: Digium will sometimes receive requests to send GPG-encrypted
mail dealing with security issues. This works somewhat poorly for
email role accounts where there are multiple recipients on a single
address. If there exists a better way to do this that doesn't involve
a lot of cust
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
>
> Thanks for pointing this out. I have located the erroneous code and have
> fixed
> it in subversion, revision 161490. The next rc of 1.6.0 will not have this
> bug.
Mark --
This bug still exists in the recently-released
hi:
When iam sending calls through sip a fake ringback tone is generated and then
call status can't be viewed (if call is ringing,busy,offline) it just rings and
rings.
Can i disable this?
Thanks in advance.
_
Windows Live™:
On Friday 09 January 2009 13:52:56 Anthony Francis wrote:
> Tilghman Lesher wrote:
> > We are entirely interested in DETERMINISTIC methods of uniqueness, not
> > random and hope-for-the-best. Given a truly random generator, it is
> > possible for the same number to come up 100 times in sequence.
On Friday 09 January 2009 13:08:41 Vieri wrote:
> --- On Fri, 1/9/09, Josiah Bryan wrote:
> > Is this right after bringing online the alias IP?
> > If so, you might try using arp-sk to broadcast an ARP
> > packet to kick-start the IP lookup...
> > http://sid.rstack.org/arp-sk/
>
> Thanks for the l
Tilghman Lesher wrote:
> On Friday 09 January 2009 01:14:37 Grey Man wrote:
>
>> On Fri, Jan 9, 2009 at 6:37 AM, Tilghman Lesher wrote:
>>
>>> I think Steve is as interested as anybody else in achieving a solution,
>>> but you're hand-waving when it comes to the establishment of a UUID.
Steve Murphy wrote:
> Sorry, I apologize for the 'uniqueID' field; I didn't invent it, or name
> it, and there is little definition for it. I think it's accidental that
> a transfer could yield two CDRs with the same uniqueID. I'm all for just
> simply dropping it. Maybe I will.
I would ask that
iknow this is not waht are you looking for but in case you need a web soft
phone
https://jain-sip-applet-phone.dev.java.net/
2009/1/9 Gustavo A Gonzalez
> Hi all! Im looking for 1ezphone to use as a web softphone but I'cant
> access to 1ezphone.com. Anyone knows what happened with this site?.
I set up a couple of PABXs this way 25 years ago. It was a little simpler then
because there was a more uniform number plan in the US back then, although most
of the industry people I talked to though I was totally crazy. It was a 300
station 3 digit extensions system. It worked well for a nu
I just found an old bug report at bugs.digium.com with exactly the same problem.
It's really too bad this bug wasn't addressed:
http://bugs.digium.com/view.php?id=7315
--- On Fri, 1/9/09, Vieri wrote:
> I'm trying to figure out how to reload iax2 (without
> breaking existing calls) so it can
On Friday 09 January 2009 01:14:37 Grey Man wrote:
> On Fri, Jan 9, 2009 at 6:37 AM, Tilghman Lesher wrote:
> > I think Steve is as interested as anybody else in achieving a solution,
> > but you're hand-waving when it comes to the establishment of a UUID.
> > There is no such construct that we ca
--- On Fri, 1/9/09, Josiah Bryan wrote:
> Is this right after bringing online the alias IP?
> If so, you might try using arp-sk to broadcast an ARP
> packet to kick-start the IP lookup...
> http://sid.rstack.org/arp-sk/
Thanks for the link.
However, it's not that the two boxes don't "see" each
Hi all! Im looking for 1ezphone to use as a web softphone but Icant access
to 1ezphone.com. Anyone knows what happened with this site?. Thanks!
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
ggonza...@despegar.com
___
-- Bandwidt
>> I want to detect brute-force password hacking attacks - thus if there
>> are too many failed login attempts for a SIP account I want to "lock"
>> this account.
>
>> Does somebody have any ideas how this could be implemented?
The usual method (I think) is to monitor the log files, and
detect re
Check out this howto: http://engineertim.com/?p=16
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Michiel van Baak" wrote:
> On 11:04, Fri 09 Jan 09, Matthew Nicholson wrote:
> > On Fri, 2009-01-09 at 16:49 +, Steve Howes wrote:
> > > On 9 Jan 2009, at 16:36, Kl
On 11:04, Fri 09 Jan 09, Matthew Nicholson wrote:
> On Fri, 2009-01-09 at 16:49 +, Steve Howes wrote:
> > On 9 Jan 2009, at 16:36, Klaus Darilion wrote:
> > > Hi!
> > >
> > > I want to detect brute-force password hacking attacks - thus if there
> > > are too many failed login attempts for a SIP
Mark Michelson a écrit :
> Benoit wrote:
>
>> Hi,
>>
>> I'm a little surprised, up until 1.4.22 my agents where using an IAX
>> channel to ZoIPer Softphone,
>> however since after the upgrade to .22 we experienced a problem with
>> hangup failure between zoiper
>> and asterisk (look like bug ht
what about realtime and dundi? one asterisk for register and then many for
the T1 cards you use dundi to route the calls if one server fail the system
will repleace it...
and a the dual core are cheap now so it is not a big issue buy 4 or five.
David
2009/1/9 Benoit
>
> Isn't anyone using this k
Olivier schrieb:
> Before diving into this, I would very pleased to know if someone could yes
> or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or beta2
> version) ?
> /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/avm_fritz.c:895: warning:
> passing argument 2 of 'request_irq' from
Benoit wrote:
> Hi,
>
> I'm a little surprised, up until 1.4.22 my agents where using an IAX
> channel to ZoIPer Softphone,
> however since after the upgrade to .22 we experienced a problem with
> hangup failure between zoiper
> and asterisk (look like bug http://bugs.digium.com/view.php?id=13184
2009/1/9 Steve Howes
> On 9 Jan 2009, at 16:36, Klaus Darilion wrote:
> > Hi!
> >
> > I want to detect brute-force password hacking attacks - thus if there
> > are too many failed login attempts for a SIP account I want to "lock"
> > this account.
> >
> > Does somebody have any ideas how this cou
On Fri, 2009-01-09 at 16:49 +, Steve Howes wrote:
> On 9 Jan 2009, at 16:36, Klaus Darilion wrote:
> > Hi!
> >
> > I want to detect brute-force password hacking attacks - thus if there
> > are too many failed login attempts for a SIP account I want to "lock"
> > this account.
> >
> > Does someb
On 9 Jan 2009, at 16:36, Klaus Darilion wrote:
> Hi!
>
> I want to detect brute-force password hacking attacks - thus if there
> are too many failed login attempts for a SIP account I want to "lock"
> this account.
>
> Does somebody have any ideas how this could be implemented?
Bad plan? Could qui
Hi!
I want to detect brute-force password hacking attacks - thus if there
are too many failed login attempts for a SIP account I want to "lock"
this account.
Does somebody have any ideas how this could be implemented?
thanks
klaus
___
-- Bandwidth a
Isn't anyone using this kind of thing http://www.red-fone.com/ for this
kind of massive HA deployment ?
there is a case study on their web site using 4 asterisk boxes and 8
red-fone T1 to Ethernet bridges to
handle 900 concurrent calls.
(I would say that using all this nice hardware on Linksys ne
Could you further clarify on this? Why is the norm shifting from 9 to 8?
-Jon
- Original Message -
From: Lyle Giese
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 09, 2009 8:45 AM
Subject: Re: [asterisk-users] Not Dialing 9
Gordon Hender
On Fri, Jan 9, 2009 at 3:08 PM, Steve Murphy wrote:
>
> By appending another string, you can guarantee it is unique across
> systems.
> So, if you use a system name, or Asterisk server name, that you yourself
> guarantee to be unique among all the asterisk servers that would
> contribute
> CDRs to
Mohit Kumar schrieb:
> I had downlaoded iaxclient-2.0.2 and complie project
> *\iaxclient-2.0.2\contrib\win\vs2005*
> **
> It gives many83 fatal and file missing error of file missing
> Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such
> file or
> directory
> d:\mohit\a
Mohit Kumar schrieb:
> I had downlaoded iaxclient-2.0.2 and complie project
> *\iaxclient-2.0.2\contrib\win\vs2005*
> **
> It gives many83 fatal and file missing error of file missing
iaxclient has its own web site and mailing list.
http://iaxclient.wiki.sourceforge.net/developers
Philipp K
Hi,
I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
made them switch to SIP
I'm pretty sure he was asking about a minimum of two boxes. 4 quad T1
cards would only be 16 x 24 = 384 lines. I agree to a point though - if
you have a service that is utilizing close to 800 lines and half of your
service suddenly bites the dust you would probably be in a world of hurt.
I t
On Fri, 2009-01-09 at 04:24 +, Grey Man wrote:
> On Fri, Jan 9, 2009 at 3:48 AM, Steve Murphy wrote:
> >
> > But, since it is timestamp based, and unique in that the final part was
> > incremented per request in the same sec, it made a great item to sort
> > on, and allowed me to implement lin
Is this right after bringing online the alias IP?
If so, you might try using arp-sk to broadcast an ARP packet to
kick-start the IP lookup...
http://sid.rstack.org/arp-sk/
-josiah
Vieri wrote:
> I'm trying to figure out how to reload iax2 (without breaking existing calls)
> so it can listen on
Gordon Henderson wrote:
> On Thu, 8 Jan 2009, Thczv F. Thczv wrote:
>
>
>> When I set up my Asterisk box at home I didn't want to have to dial 9
>> to dial off premises, so I gave all my local phones three digit
>> extensions with this format: 1[1,0]*. My thought is that there are no
>> area co
Klaus Darilion schrieb:
> Hi!
>
> I use unixodbc to connect to mysql. When the connection to the DB is
> lost (e.g. restarting the Mysql server) the reconnect takes 5 seconds:
>
> [Jan 9 13:22:15] WARNING[13899]: res_odbc.c:113
> ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting
Hi everybody,
Just wanted to let you know that the Call for Papers for AMOOCON
ends at the 25th of January 2009 ( http://www.amoocon.org/ ).
See the web site for a list of speakers.
AmooCon is the open-source telephony conference formerly known
as Asterisk-Tag.org ( http://www.asterisk-tag.org/ )
Hi!
I use unixodbc to connect to mysql. When the connection to the DB is
lost (e.g. restarting the Mysql server) the reconnect takes 5 seconds:
[Jan 9 13:22:15] WARNING[13899]: res_odbc.c:113
ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
reconnect...
[Jan 9 13:22:20] NOTIC
Hi!
I have asterisk 1.4.22 configured to register to a SIP proxy. The
problem is that if for some reason the registration fails once (e.g.
REGISTER-401-REGISTER-403), Asterisk does not try to reregister again -
I have to "sip reload" to trigger a reREGISTER.
I tried the default SIP settings an
On Thu, 8 Jan 2009, Thczv F. Thczv wrote:
> When I set up my Asterisk box at home I didn't want to have to dial 9
> to dial off premises, so I gave all my local phones three digit
> extensions with this format: 1[1,0]*. My thought is that there are no
> area codes that start with 0 or 1, so if I
I'm trying to figure out how to reload iax2 (without breaking existing calls)
so it can listen on a new IP address (like "ip addr add local ..."). This alias
IP is added/removed by a custom process (script) for clustering purposes.
The iax.conf file contains "bindaddr=0.0.0.0".
I tried a "iax2
Oops, I forgot the footer wouldn't appear in these lists:
VUC is Fridays at 12 Noon Eastern, 9AM PST, 11AM CST, 5PM UTC
http://www.voipusersconference.org
Calling: Please join Talkshoe.com and get a username and PIN so I can
see who is who.
PSTN: (724) 444-7444 22622# PIN#
SIP URI: 7463#22622#.
Hi all,
We've had Yusuf Motiwala from TringMe on the VoIP Users Conference
before when he annouced their Flash-based web phones. Now they've come
up with something that tantalizes me, VoicePHP. Sure XML is a standard
and fairly easy to implement, but not as easy as PHP, which I have
used since ver
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