date:20061109

Does anyone have any information on how to use DUNDi precaching?

Mark Spencer made a post 2 years ago where he hinted it may be possible to 
configure DUNDi such that you could centralise your DUNDi registration info by 
using precaching, instead of having each DUNDi peer meshed with every other 
one...

http://lists.digium.com/pipermail/dundi/2004-October/000189.html

However, it seems that no documentation exists for this in the known universe.

Doug.
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[asterisk-users] Re: Re: Reg errors? Other anomalies? Checkthosecapacitors!

2006-11-09 Thread Steven




Sounds like the Sony/Toshiba battery 
issue.

Contract goes to assemble company, but they 
outsource componentmanufacturing to the 
lowest bidder with no quality control.
-- -- Steven

http://www.glimasoutheast.org



  "Bruce Reeves" [EMAIL PROTECTED] 
  wrote in message news:[EMAIL PROTECTED]...Several 
  motherboard manufactures in the last 3-4 years have had capacitor problems, 
  some reached the point of leaking others began to cause problems on the 
  machine after they began to swell. Both Dell and IBM have replaced systems I 
  know of and had the onsite techs check for swollen or leaking capacitors. 
  
  On 11/8/06, Steven 
  [EMAIL PROTECTED] 
  wrote:
  Always 
take your wedding ring off when working inside the 
box!!Stevenhttp://www.glimasoutheast.org"Doug 
Crompton"  [EMAIL PROTECTED] 
wrote in message 
news:[EMAIL PROTECTED] 
The motherboard's capacitor? What is that? Since there are probably 
a hundred or more caps on the MB, how did you determine that? Was it 
burned?  Other than that, without making either capacitance or noise 
tests I can't imagine how you would make that 
assumption. Doug On Wed, 8 Nov 2006, Ronald 
Lewis wrote: Three months ago, I was experiencing all 
sorts of issues with my Asterisk box maintaining a connection to 
multiple trunks, etc. I also experienced various timing issues 
as well. In addition, Asterisk would sometimes take  almost a 
minute to fully load and register its SIP and IAX 
trunks. Puzzled, I recompiled several times. No 
result. I checked my hardware. Didn't find anything. However, I 
did overlook one thing:  * The motherboard's 
capacitor! Yep, you guessed it! It was bad. Now, I 
do not have any problems (I didn't bother replacing the 
motherboard, ended up using a spare PC). 
 "Those that sacrifice essential liberty 
to obtain a little temporary safety deserve neither liberty nor 
safety."-- Ben Franklin (1759) 
  *Doug 
Crompton* *Richboro, PA 
18954* 
*215-431-6307* 
** * [EMAIL PROTECTED]* 
* http://www.crompton.com* 
 
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Re: [asterisk-users] Problem with register command in SIP.conf

2006-11-09 Thread Anselm Martin Hoffmeister

Am Donnerstag, den 09.11.2006, 12:19 -0300 schrieb Frederico Madeira:
 I'm registering 5 lines on my asterisk box from one voip provider.
 Lines;
 
 4040.
 4040.0001
 4040.0002
 4040.0003
 4040.0004
 
 All lines is registered in 5060 port so when someone call to 4040.0001
 the call arrive on asterisk but arrive to last number registered
 4040.0004 becouse it is listening on same port as all others.
 
 How i make each number register in one different port, like
 5060,5061,5062,5063 and 5064 ?? 

That is not necessary. Have the register statement contain the extension
to be called as trailing parameter, as in

register = 5631234567:[EMAIL PROTECTED]:5060/1234567
register = 5631234568:[EMAIL PROTECTED]:5060/1234568

will call extension 1234567 or 1234678, depending on which SIP line
the call came in on.
You should have the same context for all those 4040.* of yours, and then
in that extensions.conf context, say

[ctxsipnumber]
exten = 1234567,1,Dial(SIP/sip507)
exten = 1234568,1,Dial(SIP/sip505)

or the like.

Hth
Anselm

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Re: [asterisk-users] several behind NAT

2006-11-09 Thread Todd- Asterisk

Just to report back in, the advice of the list was to not worry about it- they should work well.  I took a DSL modem with a router on it and connected both phones (Grandstream GXP2k and 101)- they did not work.  I found that I had to program in a STUN server.  I also has to set it to use a random port instead of the default- a pre-defined port (else only 1 phone would ring regardless of extension).  Now they both work well.  Does anyone see a problem with this setup?  Should I use my own STUN server? or can I continue with stun.fwdnet.net?  Also, where can I get information on provisioning?  These phones will be out of my hands soon and I'd like to be able to update the configs.  I saw a few utilities for generating the configs, but I'd like more specific info - I don't mind editing files by hand but want to know how it works. Does anyone have some resources?  thanks for all the help- this is a great list.     ToddOn Nov 6, 2006, at 10:28 AM, Todd- Asterisk wrote:I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems.  However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server.  Should I look into using STUN servers?  Will this setup be a problem?  I've read about NAT and STUN on voip-info but am looking for more information..   btw- I'm not set on Grandstream.  If you think Polycom or something can handle NAT better, then I'll use that instead.  I guess there's no IAX phones yet...  Thanks in advance.  Todd___
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Re: [asterisk-users] DTMF Corruption Problem

2006-11-09 Thread Erick Perez


On 11/9/06, mail-lists [EMAIL PROTECTED] wrote:

Erick Perez wrote:
 I can report that with asterisk 1.2.13, internal SIP calls work
 perfectly but (in my particular case) my asterisk box cannot recognize
 DTMF digits when it receives a call via our SIP provider. we are both
 using rfc2833 and I have tried relaxdtmf=yes/no

 when i use an internal sip extension and call somebody outside via my
 sip provider, dtmf is recognized.

 On 11/9/06, mail-lists [EMAIL PROTECTED] wrote:

  Also, I am not using a zaptel timer.  Could this possibly be causing
  problems with DTMF??
 I really don't know for certain but here's what I experienced: When
 calling out asterisk gives the option to allow called numbers to
 transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th
 dial string. This would very seldom work. I could hit the '#' on the
 called phone it would say 'extension' but would always reply with 'not
 valid extension'

 I recently upgraded to 1.2.12 and noticed that there was no ztdummy
 running! I compiled my own zaptel installed it, loaded the modules on
 boot and now the transfer works perfectly.

 Also: my moh wasn't working for some reason. After I installed the
 ztdummy module it works too..

 I'm not sure whether the transfer issue was fixed by using the ztdummy
 module or by the asterisk issue but my point is that you should always
 have the ztdummy module installed if possible.

 Just my .02. Hope it helps


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Erick,

Do you have ztdummy running?
What SIP provider are you using. Incoming calls work fine for me (and
always have as far as I know).



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I have a TDM400 installed. loading wctdm and not ztdummy.
centos 4.4 with kernel 2.6
My provider is not located in US...I am not lcated in the US
--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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[asterisk-users] unsubscribe

2006-11-09 Thread Adam Mattina













Adam Mattina
Networking  Systems
Support
Layer 8 Group, Inc.
585.442.
[EMAIL PROTECTED]








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RE: [asterisk-users] Question on Aastra phones and Astrisk

2006-11-09 Thread shadowym

That clarifies it!

First the stupid questions to eliminate the possibility of anything besides
the phones,

Have you connected a different make hardphone or softphone and confirmed
that works?

Have you tried a different IAX/SIP provider? 

-Original Message-
From: Matt [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, November 08, 2006 6:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk

It only happens when you go from IAX/SIP -- asterisk box -- aastra phone.
Doesn't happen PSTN -- asterisk box -- aastra phone.

The aastra people have said they believe it is a codec negotiation issue...
but the newest firmware didn't fix it send them packet dumps.

On 11/7/06, shadowym [EMAIL PROTECTED] wrote:
 Running several Aastra 9133i and 480CT phones with v1.4 firmware 
 CentOS 4.4, Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3.  Using all 
 default settings

 I have not seen that problem. I am not exactly sure we are creating 
 those exact same conditions but it sounds like standard extension use 
 to multiple incoming calls correct?  That is all we are doing plus 
 some more complicated outgoing stuff.

 -Original Message-
 From: Matt [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, November 07, 2006 5:31 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk

 *bump*  Anyone?

 On 11/6/06, Curt Shaffer [EMAIL PROTECTED] wrote:
  I wanted to add what we have both seen on traffic captures.
 
  You see Caller 1's RTP stream. Call 2 comes in and you see the 
  creation of its RTP stream. After Call 2 is put on hold the RTP 
  stream from Caller 1 disappears without a trace never to return and 
  this is when the one way audio is happening.
 
  And I also wanted to add that I am running 1.4.0 firmware for this
phone.
 
  Thanks again!
 
 
 
  -Original Message-
  From: Curt Shaffer [mailto:[EMAIL PROTECTED]
  Sent: Monday, November 06, 2006 6:58 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
 
  I'm the friend mentioned here.
 
  I am using the Aastra 480i CT. It is SIP to my PBX and IAX 
  termination from the PBX to my provider. My issue has a slight twist 
  to it but the same result. For instance his is always where as mine 
  is frequent but not
 always.
  After I got to finally see it first hand today, I had to start over 
  from Caller 1 5 times to get it to happen again.
 
  Caller 1 calls in and Person A answers. Caller 2 calls in and Person 
  B answers. Person B puts caller 2 on hold and audio drops on Caller 1.
  So Person A can hear caller 1 but caller 1 cannot hear Person A.
 
  This happens more often when Call 1 is on the handset and Call 2 is 
  on the portable or vis a vi, but this is not always the case. It 
  does happen to 1 set only but just less frequent.
 
  I have tried carrierinvite=yes and no but this does not change the
issue.
  The phones are behind a router but the external IP of the router is 
  on the same network as the * box.
 
  Thanks!
 
  Curt
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Matt
  Sent: Monday, November 06, 2006 6:35 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Question on Aastra phones and Astrisk
 
  Hi,
 Some odd behaviour here.  A friend and I were talking tonight,
  and it seems we have both seen the same problem.   We are both using
  aastra phones (I am using 9113is).We have a connection to and from
  providers via SIP and IAX.When I place a call on the local hold of
  the phone, and then pick them back up I can hear them, but they can
  not hear me.However, if I park the call, and then pick it up
  again, the audio is fine.
Tonight I tried placing a call on hold using a Sipura/Linksys 
  ATA (that is just hitting 'flash', which basically puts the call on
  local hold and starts music).The problem did not manifest itself.
 
  Has anyone else had this issue?  Do you have a fix for it?  It is an 
  astrisk issue or an aastra issue?
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[asterisk-users] asterisk and norstar

2006-11-09 Thread Gustavo Berman

Hi there!We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension.
We are in Argentina, so buying a star talk is out of the question, there is no selling of that in here.So, we want to use * as an auto attendant and voicemail for our 50 extensions.Is there anybody who has done that?
What topology do we have to use? :1) pstn line - (fxo) asterisk (fxs) - norstaror2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstaror3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk (fxo/2) - ata/2 - (
ext.321) norstaror4) pstnl line - norstar (ext. 123) - ata - (fxo) asteriskAny help please?I'm not a telephone systems specialist!Thanks!-- Gustavo BermanSysadminDepto. Informatica
Universidad Nacional del ComahueCentro Regional Universitario Bariloche
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[asterisk-users] special characters in alphanumeric extensions

2006-11-09 Thread Ricardo Carvalho


Hi all,

I use alphanumeric names as extensions in my Asterisk architecture, 
which are the username part of the e-mail of each person at my site. 
Because Asterisk was primarily built to use numeric extensions, I'm 
having some problems with people that have usernames with dots between 
letters, like john.doe.
More specifically my problem is when john.doe dials some number. 
Asterisk doesn't match his rule in extensions.conf. I have in that file 
the following line:


exten = _[0-9]./john.doe,1,Dial(SIP/[EMAIL PROTECTED],60)

When that user dials some number, Asterisk never matches his rule. This 
only happens because dots are special parameters for Asterisk. I've 
tried to put a slash \ before the dot, but nothing happens!...


Any suggestion?

Thanks in advance,
Ricardo.

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Re: [asterisk-users] Voxee lag problems ?

2006-11-09 Thread mail-lists


Vicky wrote:
Anyone having problems with voxee since last few days or is it just me 
? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 
ms latency . Most of time it is 20 ms or so but when i start sending 
traffic to them latency increases to 1000 ms or even LAGGED  ( also 
shows high in peak time even when no high latency 
). No problems with any other provider . Anyone else having same 
problem ?



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Yes.. not just the last few days either... the last few weeks
I havent' bothered to write voxee about it as their support sucks 
horribly and

it takes about a week most times for them to get back to you.

I have a voxee trunks on 2 seperate boxes and both do the same
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[asterisk-users] Bug ???

2006-11-09 Thread phil . dawson


Hi All,

I have tried everything to get callerid to work reliably but to no avail.
I have configured zapata.conf as per documentation but still only get 50%
of callerid's through.  As a test I called our system with my mobile a
number of times and only 50% get through.  I do get warnings about
polarity.  I am in the UK.

Anyone have ideas what to check?

TIA


Phil

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Re: [asterisk-users] DUNDi precache

2006-11-09 Thread Aaron Daniel

Doug,

This may help you out a little.  It's a whitepaper that JR wrote on how
to get a DUNDi cluster working with two redundant primary servers that
handle all the DUNDi legwork.  Read through it, you might get some
information you can use out of it.

http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf

On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
 Does anyone have any information on how to use DUNDi precaching?
 
 Mark Spencer made a post 2 years ago where he hinted it may be possible to 
 configure DUNDi such that you could centralise your DUNDi registration info 
 by using precaching, instead of having each DUNDi peer meshed with every 
 other one...
 
 http://lists.digium.com/pipermail/dundi/2004-October/000189.html
 
 However, it seems that no documentation exists for this in the known universe.
 
 Doug.
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] Quick Q...

2006-11-09 Thread Jay Moore

Before I make any serious gaffes, is this an acceptable place to post 
PHPAGI questions as well?  I can't seem to find a dedicated mailing list 
for it.  If not, any suggestions?


Thanks,
Jay
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Re: [asterisk-users] SRTP

2006-11-09 Thread Kristian Kielhofner


yusuf wrote:

Khaled wrote:

I  installed libsrtp can any one help me how to ingrate it with 
asterisk .to make SRTP


 


Regards


Hi,

I dont think SRTP is supported in Asterisk.  There is some work to have 
RTP over TCP, where be default its over UDP.






	SRTP has nothing to do with the transport protocol, and in fact works 
over TCP or UDP, although common sense says that it doesn't make much 
sense to transport audio (either RTP or SRTP) with TCP. :)


	There is some effort underway to support SIP over TCP, which can make 
some sense.  Note that SIP and RTP are two separate (although often 
confused) protocols that can use a mixture of UDP/TCP (TCP for the 
signaling - SIP, and UDP for the audio - RTP).  Or, %100 UDP as it is in 
Asterisk now.


--
Kristian Kielhofner


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[asterisk-users] Digium B410P

2006-11-09 Thread King Ho

Hi,

I am trying to decide which BRI card to buy and am looking at the B410P. I
have the following questions and hope that you guys can give me some advice.

1) Anyone uses the B410P with Asterisk with good or bad comment? And how
does the B410P compared to the others like the AVM or the Beronet.

2) I think the B410P is using the mISDN driver. I also read that the mISDN
driver will have problem with SMP kernel, so if I want to use the B410P, I
should not get any duo core CPU with the computer, right?  Also, if this is
true, maybe I will be better off with a card that uses the CAPI driver like
the AVM?

3) The installation guide on digium http://kb.digium.com/entry/55/131/ says
that I need at least Linux kernel 2.6.15. Does this mean that CentOS 4.4 is
not compatible with it?

Thanks in advance!!

Best Regards,

King

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[asterisk-users] A couple of new tutorials: installing * 1.4 and the Asterisk GUI

2006-11-09 Thread Lenz


Hello list,
I have prepared a couple of new tutorials you may find interesting:

- Installing an Asterisk 1.4 beta system  - at http://astrecipes.net/?n=216
- Installing the Digium's Asterisk GUI for 1.4 - at  
http://astrecipes.net/?n=217


It's nothing too complex, but you may find them interesting, especially  
the new Asterisk GUI.


Any comment is welcome - the site is a wiki, so feel free to correct any  
errors or add improvements.

l.


--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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Re: [asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!

2006-11-09 Thread Ira


At 05:00 AM 11/9/2006, you wrote:
Several motherboard manufactures in the last 3-4 years have had 
capacitor problems, some reached the point of leaking others began 
to cause problems on the machine after they began to swell. Both 
Dell and IBM have replaced systems I know of and had the onsite 
techs check for swollen or leaking capacitors.


I have an IBM where every single 470uf 25V cap on the board leaked at 
about 2.5 years. Replaced them all and it's still going strong. I 
think something went wrong in a capacitor plant somewhere a few years 
back and a whole bunch of bad ones got out in the wild.


Ira 


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RE: [asterisk-users] special characters in alphanumeric extension s

2006-11-09 Thread Colin Anderson

I use alphanumeric names as extensions in my Asterisk architecture, 
which are the username part of the e-mail of each person at my site. 
Because Asterisk was primarily built to use numeric extensions, I'm 
having some problems with people that have usernames with dots between 
letters, like john.doe.

I ran into the same problem myself and I realized while making extensions
have some meaning other than a random number is neat and can be used for
tons of other purposes (notification emails for example), it quickly becomes
unmanageable if your org changes things around a lot like the one I work
for. Consider what would happen if a user changes their email address
(happens all the time: women get married etc) then there is the added
overhead of futzing with sip.conf etc in order to accomodate the change. 

In the end, I made the SIP account number  the voicemail box number the
last 4 digits of the user's DID, which greatly simplifies things. I also set
each user's email address as a local variable when the DID is lit up by the
PSTN, so my notification emails are a snap:

exten = h,1,System(echo You hung up the call to  ${CALLERIDNAME} | mail
-s ${EMAILADDRESS})

(note that the above is just an example I pulled out of my butt, probably
would not work in real life.)

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RE: [asterisk-users] DUNDi precache

Aaron.

Thanks. JR sent me that article before it was published. He's not precaching 
registrations. He's doing something different. In his configuration, when a 
registration server gets a request for the location of a phone, it queries the 
DUNDi Lookup server, which in turn queries the other registration servers on 
it's behalf. It doesn't actually cache the registrations itself. 

According to what Mark Spencer wrote, it should be possible for this DUNDi 
Lookup server to hold, or store (ie cache) -all- phone registration info so 
that it doesn't have to query the other registration servers. 

Doug. 

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Thursday, November 09, 2006 9:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] DUNDi precache
 
 
 Doug,
 
 This may help you out a little.  It's a whitepaper that JR 
 wrote on how
 to get a DUNDi cluster working with two redundant primary servers that
 handle all the DUNDi legwork.  Read through it, you might get some
 information you can use out of it.
 
 http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf

On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
 Does anyone have any information on how to use DUNDi precaching?
 
 Mark Spencer made a post 2 years ago where he hinted it may be possible to 
 configure DUNDi such that you could centralise your DUNDi registration info 
 by using precaching, instead of having each DUNDi peer meshed with every 
 other one...
 
 http://lists.digium.com/pipermail/dundi/2004-October/000189.html
 
 However, it seems that no documentation exists for this in the known universe.
 
 Doug.
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Re: [asterisk-users] unsubscribe

2006-11-09 Thread Anthony Rodgers


Here's how to unsubscribe:

First, ask your Internet Provider to mail you an Unsubscribing Kit.
Then follow these directions.

The kit will most likely be the standard no-fault type. Depending on
requirements, System A and/or System B can be used. When operating
System A, depress lever and a plastic dalkron unsubscriber will be
dispensed through the slot immediately underneath. When you have
fastened the adhesive lip, attach connection marked by the large X
outlet hose. Twist the silver-coloured ring one inch below the
connection point until you feel it lock.

The kit is now ready for use. The Cin-Eliminator is activated by the
small switch on the lip. When securing, twist the ring back to its
initial condition, so that the two orange lines meet. Disconnect.
Place the dalkron unsubscriber in the vacuum receptacle to the rear.
Activate by pressing the blue button.

The controls for System B are located on the opposite side. The red
release switch places the Cin-Eliminator into position; it can be
adjusted manually up or down by pressing the blue manual release
button. The opening is self-adjusting. To secure after use, press the
green button, which simultaneously activates the evaporator and
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You may log off if the green exit light is on over the evaporator. If
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To use the Auto-Unsub, first undress and place all your clothes in the
clothes rack. Put on the velcro slippers located in the cabinet
immediately below. Enter the shower, taking the entire kit with
you. On the control panel to your upper right upon entering you will
see a Shower seal button. Press to activate. A green light will then
be illuminated immediately below. On the intensity knob, select the
desired setting. Now depress the Auto-Unsub activation lever. Bathe
normally.

The Auto-Unsub will automatically go off after three minutes unless
you activate the Manual off override switch by flipping it up. When
you are ready to leave, press the blue Shower seal release
button. The door will open and you may leave. Please remove the velcro
slippers and place them in their container.

If you prefer the ultrasonic log-off mode, press the indicated blue
button. When the twin panels open, pull forward by rings A  B. The
knob to the left, just below the blue light, has three settings, low,
medium or high. For normal use, the medium setting is suggested.

After these settings have been made, you can activate the device by
switching to the ON position the clearly marked red switch. If
during the unsubscribing operation you wish to change the settings,
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you.

CP

On Nov 9, 2006, at 8:01 AM, Adam Mattina wrote:



 
 
Adam Mattina
Networking  Systems Support
 Layer 8 Group, Inc.
 585.442.
[EMAIL PROTECTED]
 
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[asterisk-users] Station Voip Brazil

2006-11-09 Thread Felipe Amaral

Hi,There's anyone here who go to Estacao Voip in Brazil???http://www.estacaovoip.com.br/I was think to goAnyone here ??
-- Felipe AmaralVento Livre Internet
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Re: [asterisk-users] DUNDi precache

2006-11-09 Thread Bruce Reeves

Doug,JR's example does cache but only for a very short time, like 5 seconds, so that if the device registers else where then the lookup is able to find it. You can change the cache time to the default hour or what ever you want. As far as precahe, I know it is a dundi cli command and you could probably script connecting to the cli and precacheing an extension, but in my case the lookups are under 70 ms so I don't notice a hughe hit on performance when a lookup is done. I know you are trying to use this on a much larger scale then I do.
On 11/9/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Aaron.Thanks. JR sent me that article before it was published. He's not precaching registrations. He's doing something different. In his configuration, when a registration server gets a request for the location of a phone, it queries the DUNDi Lookup server, which in turn queries the other registration servers on it's behalf. It doesn't actually cache the registrations itself.
According to what Mark Spencer wrote, it should be possible for this DUNDi Lookup server to hold, or store (ie cache) -all- phone registration info so that it doesn't have to query the other registration servers.
Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: Thursday, November 09, 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi precache Doug, This may help you out a little.It's a whitepaper that JR wrote on how to get a DUNDi cluster working with two redundant primary servers that
handle all the DUNDi legwork.Read through it, you might get some information you can use out of it. http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
ith%20a%20Cluster%20of%20Asterisk%20Servers.pdfOn Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one...
http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no documentation exists for this in the known universe.
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Re: [asterisk-users] DUNDi precache

2006-11-09 Thread Michiel van Baak

On 10:16, Thu 09 Nov 06, Douglas Garstang wrote:
 Aaron.
 
 Thanks. JR sent me that article before it was published. He's not precaching 
 registrations. He's doing something different. In his configuration, when a 
 registration server gets a request for the location of a phone, it queries 
 the DUNDi Lookup server, which in turn queries the other registration servers 
 on it's behalf. It doesn't actually cache the registrations itself. 
 
 According to what Mark Spencer wrote, it should be possible for this DUNDi 
 Lookup server to hold, or store (ie cache) -all- phone registration info so 
 that it doesn't have to query the other registration servers. 
 
 Doug. 

Hi,

The cachetime is set to 5 seconds in this pdf document.
If you dont specify it it will be set to 1 hour.
Now there is your caching (right ?)
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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RE: [asterisk-users] Auto record a call?

2006-11-09 Thread Ed Nuñez

This is how I'm able to record my outbound calls, hope this helps you.

exten = _407NXX,1,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}-OUT)
exten = _407NXX,n,Monitor(wav,${CALLFILENAME},m)
exten = _407NXX,n,Dial(ZAP/g1/1${EXTEN:0})
exten = _407NXX,n,Congestion



Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Wednesday, November 08, 2006 9:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Auto record a call?

I have a debugging scenario where I wish to record the entire call.  The
call is establish via a .call file.  I can't seem to get Monitor to do
anything.  My dialplan looks like this:

[dialout]
exten = s,1,DigitTimeout,1
exten = s,n,ResponseTimeout,10
exten = s,n,Answer
exten = s,n,Monitor(wav,/tmp/test)
.
.
.


The file test.wav never shows up.  Am I doing something wrong, or
possibly there is a better way to accomplish this?

Thanks,
MC
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RE: [asterisk-users] DUNDi precache

2006-11-09 Thread Aaron Daniel

Why would you want to do that?  Defeats the purpose of *having* the
DUNDi protocol.  Why not just program the extensions in at regular
intervals or something?

On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote:
 Aaron.
 
 Thanks. JR sent me that article before it was published. He's not precaching 
 registrations. He's doing something different. In his configuration, when a 
 registration server gets a request for the location of a phone, it queries 
 the DUNDi Lookup server, which in turn queries the other registration servers 
 on it's behalf. It doesn't actually cache the registrations itself. 
 
 According to what Mark Spencer wrote, it should be possible for this DUNDi 
 Lookup server to hold, or store (ie cache) -all- phone registration info so 
 that it doesn't have to query the other registration servers. 
 
 Doug. 
 
  -Original Message-
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]
  Sent: Thursday, November 09, 2006 9:55 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] DUNDi precache
  
  
  Doug,
  
  This may help you out a little.  It's a whitepaper that JR 
  wrote on how
  to get a DUNDi cluster working with two redundant primary servers that
  handle all the DUNDi legwork.  Read through it, you might get some
  information you can use out of it.
  
  http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
 ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf
 
 On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
  Does anyone have any information on how to use DUNDi precaching?
  
  Mark Spencer made a post 2 years ago where he hinted it may be possible to 
  configure DUNDi such that you could centralise your DUNDi registration info 
  by using precaching, instead of having each DUNDi peer meshed with every 
  other one...
  
  http://lists.digium.com/pipermail/dundi/2004-October/000189.html
  
  However, it seems that no documentation exists for this in the known 
  universe.
  
  Doug.
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 http://lists.digium.com/mailman/listinfo/asterisk-users

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Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] New Asterisk 1.4 GUI

2006-11-09 Thread Curt Shaffer









I was just going to test out the new Asterisk 1.4 GUI. I
downloaded it from source make;make install. I added my http.conf and modified
manager.conf. I restarted Asterisk and did a make checkconfig and it says
everything looks good. But I notice that the port 8088 is not listening when I
do a netstat. Am I missing a step here somewhere?



Thanks



Curt






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[asterisk-users] wip5000 roaming

2006-11-09 Thread Altus Snyman









Good day all

I cant get my WIP 5000 to roam 100%

I have 2 access points, different SSIs

I make a config1 and config2 on the phone, each for the different
SSIDs(A  B)

Im standing next to A and I walk to B, butthe phone
does not want to change its signal to B, it still keeps the bad signal from A

If I power A down, it will switch to B, if I switch A back
on and go stand next to it, it will still keep Bs signal

We got some wireless specialists in and they set up
WDS for us, in other words, you add 1 SSID for both access point

IT works for windows, but not for the phone!

Can anyone help, or give a bit more explanation on the roaming
settings on the webconfig



Try RxLevel(-103~0)

PreRoaming Enable RxLevel(-103~0)

Try Over TxError Count(0~1)

Try Over RxError Count(0~1)

Level Diff Higher Than Curr Site(0~255)

Use Refresh PreRoaming

Enable PreRoaming On Association

PreRoaming Mode

PreRoaming Refresh Interval(0:Disable, 0~3600)





Thanks

Altus








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Re: [asterisk-users] New Asterisk 1.4 GUI

2006-11-09 Thread Lenz



Could it be you did not bind it correctly in http.conf? Something similar  
happened to me today while I was doing the same thing.


Try:
enabled=yes
enablestatic=yes
bindaddr=0.0.0.0

Hope this helps
l.

On Thu, 09 Nov 2006 19:28:17 +0100, Curt Shaffer [EMAIL PROTECTED]  
wrote:


I was just going to test out the new Asterisk 1.4 GUI. I downloaded it  
from
source make;make install. I added my http.conf and modified  
manager.conf. I
restarted Asterisk and did a make checkconfig and it says everything  
looks
good. But I notice that the port 8088 is not listening when I do a  
netstat.

Am I missing a step here somewhere?


Thanks


Curt





--
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http://queuemetrics.loway.it
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RE: [asterisk-users] DUNDi precache

If you have a large number of servers, a mesh relationship may not scale well, 
and maintaining relationships between all the servers is difficult for 
starters. It also cuts down on the physical distance to perform queries if you 
have a centralised DUNDi server, rather than having to query the peers on 
remote sites.

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Thursday, November 09, 2006 11:04 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] DUNDi precache

 Why would you want to do that?  Defeats the purpose of *having* the
 DUNDi protocol.  Why not just program the extensions in at regular
 intervals or something?

 On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote:
  Aaron.

  Thanks. JR sent me that article before it was published. 
 He's not precaching registrations. He's doing something 
 different. In his configuration, when a registration server 
 gets a request for the location of a phone, it queries the 
 DUNDi Lookup server, which in turn queries the other 
 registration servers on it's behalf. It doesn't actually 
 cache the registrations itself. 

  According to what Mark Spencer wrote, it should be possible 
 for this DUNDi Lookup server to hold, or store (ie cache) 
 -all- phone registration info so that it doesn't have to 
 query the other registration servers. 

  Doug. 

   -Original Message-
   From: Aaron Daniel [mailto:[EMAIL PROTECTED]
   Sent: Thursday, November 09, 2006 9:55 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] DUNDi precache

   Doug,

   This may help you out a little.  It's a whitepaper that JR 
   wrote on how
   to get a DUNDi cluster working with two redundant primary 
 servers that
   handle all the DUNDi legwork.  Read through it, you might get some
   information you can use out of it.

   http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
  ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf

  On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
   Does anyone have any information on how to use DUNDi precaching?

   Mark Spencer made a post 2 years ago where he hinted it 
 may be possible to configure DUNDi such that you could 
 centralise your DUNDi registration info by using precaching, 
 instead of having each DUNDi peer meshed with every other one...

   http://lists.digium.com/pipermail/dundi/2004-October/000189.html

   However, it seems that no documentation exists for this 
 in the known universe.

   Doug.
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 -- 
 Aaron Daniel
 Computer Systems Technician
 Sam Houston State University
 [EMAIL PROTECTED]
 (936) 294-4198

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Re: [asterisk-users] wip5000 roaming

2006-11-09 Thread Andrew Joakimsen

Disable WDS but set all the AP to the same channel and same SSID and then make sure they are connected to the same LAN (IE: no NAT on the AP). Are you using encryption?Something like:
Try RxLevel -60PreRoaming Enable RxLevel -75Try over TxErrcnt 15Try Over RxError Count 10Play with the PreRoaming mode, see if it does help? It should however you could notice a drop in battery life.
Would be a good place to start with your settings, adjust from there. I would like to hear your results with these phones, is everything working great besides the roaming?
On 11/9/06, Altus Snyman [EMAIL PROTECTED] wrote:
















Good day all

I cant get my WIP 5000 to roam 100%

I have 2 access points, different SSI's

I make a config1 and config2 on the phone, each for the different
SSID's(A  B)

Im standing next to A and I walk to B, but…the phone
does not want to change its signal to B, it still keeps the bad signal from A

If I power A down, it will switch to B, if I switch A back
on and go stand next to it, it will still keep B's signal

We got some wireless specialist's in and they set up
WDS for us, in other words, you add 1 SSID for both access point

IT works for windows, but not for the phone!

Can anyone help, or give a bit more explanation on the roaming
settings on the webconfig



Try RxLevel(-103~0)

PreRoaming Enable RxLevel(-103~0)

Try Over TxError Count(0~1)

Try Over RxError Count(0~1)

Level Diff Higher Than Curr Site(0~255)

Use Refresh PreRoaming

Enable PreRoaming On Association

PreRoaming Mode

PreRoaming Refresh Interval(0:Disable, 0~3600)





Thanks

Altus









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RE: [asterisk-users] DUNDi precache

Bruce,

After
thinking about it a bit, I can see how setting the cache time to some value
higher than 0 could be effective. However, I'm trying to figure out what
benefits a 'central' DUNDi cache server provides over a completely distributed
architecture. If you have three asterisk boxes, and each peers with the other
two, AND the cache time is set, to say, an hour, when an asterisk server locates
a phone, it will store that it in it's own cache anyway. About the only
advantage I can see is is in managability. You no longer need to maintain peer
relationships between every asterisk box, only between every Asterisk box and
the central DUNDi cache server.

Doug.

-Original Message-From: Bruce Reeves
[mailto:[EMAIL PROTECTED]Sent: Thursday, November 09,
2006 10:56 AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] DUNDi
precacheDoug,JR's example does cache but only for
a very short time, like 5 seconds, so that if the device registers else where
then the lookup is able to find it. You can change the cache time to the
default hour or what ever you want. As far as precahe, I know it is a dundi
cli command and you could probably script connecting to the cli and
precacheing an extension, but in my case the lookups are under 70 ms so I
don't notice a hughe hit on performance when a lookup is done. I know you are
trying to use this on a much larger scale then I do.
On 11/9/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Aaron.Thanks.
JR sent me that article before it was published. He's not precaching
registrations. He's doing something different. In his configuration, when a
registration server gets a request for the location of a phone, it queries
the DUNDi Lookup server, which in turn queries the other registration
servers on it's behalf. It doesn't actually cache the registrations itself.
According to what Mark Spencer wrote, it should be possible for this
DUNDi Lookup server to hold, or store (ie cache) -all- phone registration
info so that it doesn't have to query the other registration
servers.Doug. -Original Message- From:
Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: Thursday,
November 09, 2006 9:55 AM To: Asterisk Users Mailing List -
Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi
precache Doug, This may help you out
a little.It's a whitepaper that JR wrote on how
to get a DUNDi cluster working with two redundant primary servers that
handle all the DUNDi legwork.Read through it, you might
get some information you can use out of it. http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
ith%20a%20Cluster%20of%20Asterisk%20Servers.pdfOn Thu,
2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have
any information on how to use DUNDi precaching? Mark Spencer
made a post 2 years ago where he hinted it may be possible to configure
DUNDi such that you could centralise your DUNDi registration info by using
precaching, instead of having each DUNDi peer meshed with every other one...
http://lists.digium.com/pipermail/dundi/2004-October/000189.html
However, it seems that no documentation exists for this in the known
universe. Doug.
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RE: [asterisk-users] wip5000 roaming

2006-11-09 Thread Altus Snyman









Everything is working beside roaming

Yes im using encryption, should I turn it
off, or uses the same wep key, and same ssid

Should I then also just add 1 config with
1 access point , not 2?











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Thursday, November 09, 2006
8:53 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
wip5000 roaming





Disable WDS but set all
the AP to the same channel and same SSID and then make sure they are connected
to the same LAN (IE: no NAT on the AP). Are you using encryption?

Something like:

Try RxLevel -60
PreRoaming Enable RxLevel -75
Try over TxErrcnt 15
Try Over RxError Count 10

Play with the PreRoaming mode, see if it does help? It should however you could
notice a drop in battery life. 

Would be a good place to start with your settings, adjust from there. I
would like to hear your results with these phones, is everything working great
besides the roaming?



On 11/9/06, Altus
Snyman [EMAIL PROTECTED]
wrote:





Good
day all

I
cant get my WIP 5000 to roam 100%

I
have 2 access points, different SSI's

I
make a config1 and config2 on the phone, each for the different SSID's(A 
B)

Im
standing next to A and I walk to B, butthe phone does not want to change its
signal to B, it still keeps the bad signal from A

If
I power A down, it will switch to B, if I switch A back on and go stand next to
it, it will still keep B's signal

We
got some wireless specialist's in and they set up WDS for us, in other words,
you add 1 SSID for both access point

IT
works for windows, but not for the phone!

Can
anyone help, or give a bit more explanation on the roaming settings on the
webconfig



Try
RxLevel(-103~0)

PreRoaming
Enable RxLevel(-103~0)

Try Over
TxError Count(0~1)

Try Over
RxError Count(0~1)

Level
Diff Higher Than Curr Site(0~255)

Use
Refresh PreRoaming

Enable
PreRoaming On Association

PreRoaming
Mode

PreRoaming
Refresh Interval(0:Disable,
0~3600)





Thanks

Altus








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Re: [asterisk-users] Voxee lag problems ?

2006-11-09 Thread marcelobiz


Hi Vicky,

I used to use their termination services, but I had the same problems ... It is impossible to work with that latency ... A lot of gaps and dead spots ...

If you know ... I'm looking for a good termination provider that I can use the combination IAX/iLBC ... If you know some .. can you please tell me ?

Thanks,

-- Original message -- From: Vicky [EMAIL PROTECTED] Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ).Noproblemswithanyotherprovider . Anyone else having same problem ? 
---BeginMessage---
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RE: [asterisk-users] DUNDi precache

I also just realised a distinct advantage of the precache model.

Lets say you have a central DUNDi cache server. He has in his cache the 
knowledge that appearance 2944093 is registered to pbx1 for the next hour. If 
pbx1 where to crash, then for the next hour, calls to 2944093 would fail. 

However, in the precache model, when the phone registers to an Asterisk box, 
the Asterisk box immediately precaches the information to the central DUNDi 
server, who maintains this information until it's updated. If pbx1 where to 
crash, and the phone failed over to pbx2, pbx2 would then send updated 
registration information to the DUNDi precache server, and thus calls would not 
fail for an hour.

Douglas.

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Thursday, November 09, 2006 11:04 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] DUNDi precache
 
 
 Why would you want to do that?  Defeats the purpose of *having* the
 DUNDi protocol.  Why not just program the extensions in at regular
 intervals or something?
 
 On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote:
  Aaron.
  
  Thanks. JR sent me that article before it was published. 
 He's not precaching registrations. He's doing something 
 different. In his configuration, when a registration server 
 gets a request for the location of a phone, it queries the 
 DUNDi Lookup server, which in turn queries the other 
 registration servers on it's behalf. It doesn't actually 
 cache the registrations itself. 
  
  According to what Mark Spencer wrote, it should be possible 
 for this DUNDi Lookup server to hold, or store (ie cache) 
 -all- phone registration info so that it doesn't have to 
 query the other registration servers. 
  
  Doug. 
  
   -Original Message-
   From: Aaron Daniel [mailto:[EMAIL PROTECTED]
   Sent: Thursday, November 09, 2006 9:55 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] DUNDi precache
   
   
   Doug,
   
   This may help you out a little.  It's a whitepaper that JR 
   wrote on how
   to get a DUNDi cluster working with two redundant primary 
 servers that
   handle all the DUNDi legwork.  Read through it, you might get some
   information you can use out of it.
   
   http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
  ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf
  
  On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
   Does anyone have any information on how to use DUNDi precaching?
   
   Mark Spencer made a post 2 years ago where he hinted it 
 may be possible to configure DUNDi such that you could 
 centralise your DUNDi registration info by using precaching, 
 instead of having each DUNDi peer meshed with every other one...
   
   http://lists.digium.com/pipermail/dundi/2004-October/000189.html
   
   However, it seems that no documentation exists for this 
 in the known universe.
   
   Doug.
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 -- 
 Aaron Daniel
 Computer Systems Technician
 Sam Houston State University
 [EMAIL PROTECTED]
 (936) 294-4198
 
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[asterisk-users] register suddenly fails

2006-11-09 Thread Norbert Zawodsky

Hi everybody,

I've got a very strange problem:

As far as I remember I didn't change anything on my Asterisk side. I
have 2 SIP providers to which I can place outbound calls.

Today I noticed that outbound calls to provider inode fail (and
inbound from this provider too). On the CLI I get every 20 seconds
following messages:

Nov  9 20:01:07 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #1)
Nov  9 20:01:07 WARNING[952]: chan_sip.c:1998 create_addr: No such host:
voip.inode.at
Nov  9 20:01:07 WARNING[952]: chan_sip.c:5505 transmit_register:
Probably a DNS error for registration to [EMAIL PROTECTED], trying
REGISTER again (after 20 seconds)
Nov  9 20:01:27 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #2)
Nov  9 20:01:28 WARNING[952]: chan_sip.c:1998 create_addr: No such host:
voip.inode.at
Nov  9 20:01:28 WARNING[952]: chan_sip.c:5505 transmit_register:
Probably a DNS error for registration to [EMAIL PROTECTED], trying
REGISTER again (after 20 seconds

DNS lookup works:

[EMAIL PROTECTED]:~# ping voip.inode.at
PING voip.inode.at (212.41.253.181) 56(84) bytes of data.
64 bytes from 212-41-253-181.inhouse-line.inode.at (212.41.253.181):
icmp_seq=1 ttl=60 time=15.3 ms
64 bytes from 212-41-253-181.inhouse-line.inode.at (212.41.253.181):
icmp_seq=2 ttl=60 time=15.9 ms

--- voip.inode.at ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1001ms
rtt min/avg/max/mdev = 15.375/15.669/15.963/0.294 ms

Since I am sure that I didn't change anything within the last week, I
called inode support. But they said, that they didn't change anything
either.
Next I tried was a 'SIP RELOAD' which produced following output:

asterina*CLI sip reload
Nov  9 20:02:48 WARNING[952]: acl.c:244 ast_get_ip_or_srv: Unable to
lookup 'h.  }
'
Nov  9 20:02:48 WARNING[952]: chan_sip.c:1998 create_addr: No such host:
voip.inode.at
Nov  9 20:02:48 WARNING[952]: chan_sip.c:5505 transmit_register:
Probably a DNS error for registration to [EMAIL PROTECTED], trying
REGISTER again (after 20 seconds)
Nov  9 20:03:08 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #1)
Nov  9 20:03:08 WARNING[952]: chan_sip.c:1998 create_addr: No such host:
voip.inode.at
Nov  9 20:03:08 WARNING[952]: chan_sip.c:5505 transmit_register:
Probably a DNS error for registration to [EMAIL PROTECTED], trying
REGISTER again (after 20 seconds)
asterina*CLI

Now, what makes me wonder ist the first line after the reload which says
Unable to lookup 'h.  }.

Anybody of you got any idea ??

Norbert

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Re: [asterisk-users] SRTP

2006-11-09 Thread Patrick

On Thu, 2006-11-09 at 14:05 +0200, yusuf wrote:
 Khaled wrote:
  I  installed libsrtp can any one help me how to ingrate it with asterisk 
  .to make SRTP  
  
  Regards
  
 Hi,
 
 I dont think SRTP is supported in Asterisk.  There is some work to have RTP 
 over TCP, where be 
 default its over UDP.

The SRPMs at http://www.laimbock.com/asterisk/ can be built with an
optional SIP TCP patch.

Regards,
Patrick

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[asterisk-users] Harris 20-20

2006-11-09 Thread Alyed Tzompa


		Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is that when dialing to the Harris PBX it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I dial an extension,
say 100, and play it a prompt as soon as it is picked up, the promt is
beign played as soon as it reaches the Harris, eventhough the given
extension can still be ringing. If I let it ring and then pick up this
extension I only hear the prompt in the middle (or as far as it went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
		

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Re: [asterisk-users] DUNDi precache

2006-11-09 Thread Bruce Reeves

This exact problem is solved by the short cache time, this is one of the reasons behind the low cache time in the white paper by JR.Your example is correct also, but your are expecting pbx2 to push the information to the server whereas JR has the central server pull the information. Both work, but cache is automatic and the traffic is as needed not per registration.
As to your comments about a central cache server, I make use of it in our deployment over 8 sites connected via a WAN. The central server model cuts down the complexity of the configurations on remote sites. I am working on putting a second cache server in a remote site for redundancy. Again, even over the wan links the lookup time from server A to the central to server B and back is 70 to 160 ms, plenty fast to not cause a pause in the dialing of a call.
On 11/9/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I also just realised a distinct advantage of the precache model.Lets say you have a central DUNDi cache server. He has in his cache the knowledge that appearance 2944093 is registered to pbx1 for the next hour. If pbx1 where to crash, then for the next hour, calls to 2944093 would fail.
However, in the precache model, when the phone registers to an Asterisk box, the Asterisk box immediately precaches the information to the central DUNDi server, who maintains this information until it's updated. If pbx1 where to crash, and the phone failed over to pbx2, pbx2 would then send updated registration information to the DUNDi precache server, and thus calls would not fail for an hour.
Douglas. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: Thursday, November 09, 2006 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] DUNDi precache Why would you want to do that?Defeats the purpose of *having* the DUNDi protocol.Why not just program the extensions in at regular
intervals or something? On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote: Aaron. Thanks. JR sent me that article before it was published. He's not precaching registrations. He's doing something
different. In his configuration, when a registration server gets a request for the location of a phone, it queries the DUNDi Lookup server, which in turn queries the other registration servers on it's behalf. It doesn't actually
cache the registrations itself. According to what Mark Spencer wrote, it should be possible for this DUNDi Lookup server to hold, or store (ie cache) -all- phone registration info so that it doesn't have to
query the other registration servers. Doug.-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED]
] Sent: Thursday, November 09, 2006 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi precache
Doug, This may help you out a little.It's a whitepaper that JR wrote on how to get a DUNDi cluster working with two redundant primary
servers that handle all the DUNDi legwork.Read through it, you might get some information you can use out of it.
http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote: Does anyone have any information on how to use DUNDi precaching?
Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching,
instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html
However, it seems that no documentation exists for this in the known universe. Doug. ___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
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[asterisk-users] Re: DTMF Corruption Problem

2006-11-09 Thread Justin Tunney


UPDATE:

I've installed ztdummy and have chan_zap.so loaded in to the system so
Asterisk can use psuedo zap channels or whatever it does for timing.
I also specified relaxdtmf=yes in sip.conf.

DTMF is still completely awful.  Digits are still getting doubled up
and my IVR is still nearly impossible to use.

I'd like to note that I'm using Bandwidth.Com (A Level3 reseller) for
SIP VoIP service.  I did not experience this problem before with my
old VoIP provider Gafachi.

Does anyone have some tips on trying to debug this further?  Are there
any commands I can put in to tethereal that will give me debug
information on RFC2833 in the RTP stream?

Thanks,

Justin Tunney
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[asterisk-users] porting numbers away from packet 8?

2006-11-09 Thread Dean Collins









Does anyone know if its possible to port a number AWAY
from packet8?



Ive been with them for 2 years and really want to
move to an IAX based service so I can have more than 1 incoming line at a time.









Cheers,



Dean










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Re: [asterisk-users] Re: DTMF Corruption Problem


Justin Tunney wrote:

UPDATE:

I've installed ztdummy and have chan_zap.so loaded in to the system so
Asterisk can use psuedo zap channels or whatever it does for timing.
I also specified relaxdtmf=yes in sip.conf.

DTMF is still completely awful.  Digits are still getting doubled up
and my IVR is still nearly impossible to use.

I'd like to note that I'm using Bandwidth.Com (A Level3 reseller) for
SIP VoIP service.  I did not experience this problem before with my
old VoIP provider Gafachi.

Does anyone have some tips on trying to debug this further?  Are there
any commands I can put in to tethereal that will give me debug
information on RFC2833 in the RTP stream?


relaxdtmf does not apply to out of band DTMF.  With Level 3 only INBAND 
DTMF will work and the codec must be ulaw.

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Re: [asterisk-users] wip5000 roaming

2006-11-09 Thread Andrew Joakimsen

Yes, same WEP key, SSID and channel for all the AP.On 11/9/06, Altus Snyman [EMAIL PROTECTED] wrote:
















Everything is working beside roaming

Yes im using encryption, should I turn it
off, or uses the same wep key, and same ssid

Should I then also just add 1 config with
1 access point , not 2?











From: 
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Andrew Joakimsen
Sent: Thursday, November 09, 2006
8:53 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
wip5000 roaming





Disable WDS but set all
the AP to the same channel and same SSID and then make sure they are connected
to the same LAN (IE: no NAT on the AP). Are you using encryption?

Something like:

Try RxLevel -60
PreRoaming Enable RxLevel -75
Try over TxErrcnt 15
Try Over RxError Count 10

Play with the PreRoaming mode, see if it does help? It should however you could
notice a drop in battery life. 

Would be a good place to start with your settings, adjust from there. I
would like to hear your results with these phones, is everything working great
besides the roaming?



On 11/9/06, Altus
Snyman [EMAIL PROTECTED]
wrote:





Good
day all

I
cant get my WIP 5000 to roam 100%

I
have 2 access points, different SSI's

I
make a config1 and config2 on the phone, each for the different SSID's(A 
B)

Im
standing next to A and I walk to B, but…the phone does not want to change its
signal to B, it still keeps the bad signal from A

If
I power A down, it will switch to B, if I switch A back on and go stand next to
it, it will still keep B's signal

We
got some wireless specialist's in and they set up WDS for us, in other words,
you add 1 SSID for both access point

IT
works for windows, but not for the phone!

Can
anyone help, or give a bit more explanation on the roaming settings on the
webconfig



Try
RxLevel(-103~0)

PreRoaming
Enable RxLevel(-103~0)

Try Over
TxError Count(0~1)

Try Over
RxError Count(0~1)

Level
Diff Higher Than Curr Site(0~255)

Use
Refresh PreRoaming

Enable
PreRoaming On Association

PreRoaming
Mode

PreRoaming
Refresh Interval(0:Disable,
0~3600)





Thanks

Altus








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Re: [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer

2006-11-09 Thread Andres


Can you try a sip.conf entry with the port= parameter as well.

For example:

[lucent]
type=friend
host=ip of lucent box
port=5060
insecure=port,invite
context=default

Your INVITE header is including the port, and maybe Asterisk is having 
trouble matching the sip.conf entry.





So the insecure=port,invite option should also include an
insecure=user option to disregard any user info in the invite.  Is
there is another mechanism in Asterisk to disregard any user info from
an invite?

Thanks.

JR




--
Andres
Technical Support
http://www.telesip.net

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RE: [asterisk-users] DUNDi precache

2006-11-09 Thread Aaron Daniel

I still don't see a reason to use it.  If you want immediate information
about phones even in the event of a catastrophic failure, bypass the
cache altogether (that's what we do) and have it do a lookup every time.
Also, set your lookup time to an acceptable value in the event that the
primary DUNDi servers don't find the phone.  I think ours is extremely
low since we've only got a small number of servers.

From the stats that I got from JR's talk on DUNDi at Astricon, seems to
me the overhead of doing the DUNDi lookups are almost nil, so personally
I think caching is pointless in a highly available environment.

On Thu, 2006-11-09 at 12:19 -0700, Douglas Garstang wrote:
 I also just realised a distinct advantage of the precache model.
 
 Lets say you have a central DUNDi cache server. He has in his cache the 
 knowledge that appearance 2944093 is registered to pbx1 for the next hour. If 
 pbx1 where to crash, then for the next hour, calls to 2944093 would fail. 
 
 However, in the precache model, when the phone registers to an Asterisk box, 
 the Asterisk box immediately precaches the information to the central DUNDi 
 server, who maintains this information until it's updated. If pbx1 where to 
 crash, and the phone failed over to pbx2, pbx2 would then send updated 
 registration information to the DUNDi precache server, and thus calls would 
 not fail for an hour.
 
 Douglas.
 
  -Original Message-
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]
  Sent: Thursday, November 09, 2006 11:04 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [asterisk-users] DUNDi precache
  
  
  Why would you want to do that?  Defeats the purpose of *having* the
  DUNDi protocol.  Why not just program the extensions in at regular
  intervals or something?
  
  On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote:
   Aaron.
   
   Thanks. JR sent me that article before it was published. 
  He's not precaching registrations. He's doing something 
  different. In his configuration, when a registration server 
  gets a request for the location of a phone, it queries the 
  DUNDi Lookup server, which in turn queries the other 
  registration servers on it's behalf. It doesn't actually 
  cache the registrations itself. 
   
   According to what Mark Spencer wrote, it should be possible 
  for this DUNDi Lookup server to hold, or store (ie cache) 
  -all- phone registration info so that it doesn't have to 
  query the other registration servers. 
   
   Doug. 
   
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 09, 2006 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi precache


Doug,

This may help you out a little.  It's a whitepaper that JR 
wrote on how
to get a DUNDi cluster working with two redundant primary 
  servers that
handle all the DUNDi legwork.  Read through it, you might get some
information you can use out of it.

http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20w
   ith%20a%20Cluster%20of%20Asterisk%20Servers.pdf
   
   On Thu, 2006-11-09 at 08:36 -0700, Douglas Garstang wrote:
Does anyone have any information on how to use DUNDi precaching?

Mark Spencer made a post 2 years ago where he hinted it 
  may be possible to configure DUNDi such that you could 
  centralise your DUNDi registration info by using precaching, 
  instead of having each DUNDi peer meshed with every other one...

http://lists.digium.com/pipermail/dundi/2004-October/000189.html

However, it seems that no documentation exists for this 
  in the known universe.

Doug.
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  -- 
  Aaron Daniel
  Computer Systems Technician
  Sam Houston State University
  [EMAIL PROTECTED]
  (936) 294-4198
  
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] Zaptel 1.2.11 released

2006-11-09 Thread Asterisk Development Team

The Asterisk Development Team is pleased to announce the release of
version 1.2.11 of Zaptel.

This release includes a small number of fixes, primarily to support
recently updated hardware products from Digium. It also contains a very
large XPP driver update from Xorcom for their Zaptel-compatible products.

Thanks for supporting Asterisk and Zaptel!
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[asterisk-users] Modprobe Zaptel

2006-11-09 Thread Julian Varanini




Hi,

Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found"

Thanks

Julian
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Re: [asterisk-users] Station Voip Brazil

2006-11-09 Thread Kristian Kielhofner


Felipe Amaral wrote:

Hi,

There's anyone here who go to Estacao Voip in Brazil???

http://www.estacaovoip.com.br/


I was think to go


Anyone here ??


--
Felipe Amaral
Vento Livre Internet



Felipe,

I will be there, and so will Mark :).

--
Kristian Kielhofner
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[asterisk-users] Mitel 5224 Asterisk Distinctive Ring -- Anyone have it working?

2006-11-09 Thread Christopher Aloi


Hello List,

So I have a few MiTel 5224 IP phones running in SIP mode.  Per the
phones documentation they honor SIP distinctive ring tones.  I am able
to send the correct ALERT_INFO message in an invite from Asterisk to
the phone, but I don't know what ring tone to call.  From the reading
I've done the syntax is:

[3155791234]
exten = s,1,Set(_ALERT_INFO=Ring 8)
exten = s,2,Answer()
exten = s,3,Set(CALLERID(name)=FOOBAR ${CALLERIDNUM})
exten = s,4,Dial(SIP/3155791234,,r)

When I place a call to 3155791234 I see the following from Asterisk:

   -- Executing Answer(SIP/69.67.248.00-b7b1eb08, ) in new stack
   -- Executing Goto(SIP/69.67.248.00-b7b1eb08, 3155791234|s|1)
in new stack
   -- Goto (3155791234,s,1)
   -- Executing Set(SIP/69.67.248.00-b7b1eb08, _ALERT_INFO=Ring
8) in new stack
   -- Executing Answer(SIP/69.67.248.00-b7b1eb08, ) in new stack
   -- Executing Set(SIP/69.67.248.00-b7b1eb08,
CALLERID(name)=FOOBAR 311234) in new stack
   -- Executing Dial(SIP/69.67.248.00-b7b1eb08,
SIP/3155791234||r) in new stack

And the SIP invite looks like this:

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 69.67.250.00:5060;branch=z9hG4bK184f04fd;rport
From: FOOBAR 311234 sip:[EMAIL PROTECTED];tag=as26559e24
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Thu, 09 Nov 2006 23:19:28 GMT
Alert-Info: Ring 8 --- HERE IS THE ALERT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242

So it seems I have everything setup correctly.

The problem is I don't know how the phone refers to it's ring tones, I
used Ring 8 because the phone uses Ring 1-16 in the user web
interface.

Anyone have any thoughts?



--
--
Christopher T Aloi
--
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Re: [asterisk-users] Modprobe Zaptel


Julian Varanini wrote:

Hi,
 
Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get module zaptel not found


You need to edit /usr/src/linux/Makefile to remove make the EXTRAVERSION 
variable equal to -12 rather than the -12somethingsomethingsomething 
it is now.  No need to recompile the kernel, just change the make file 
and recompile and reinstall zaptel.

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RE: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer

2006-11-09 Thread Dean Collins









I know that on my blog I have a flash
player which is just html generated from xml feeds.

http://deancollinsblog.blogspot.com/



Can a html web page be auto generated from
within the Asterisk voicemail module and be sent to an email? 



What about auto generating a html email
with a player embeded in the html email? One of the companies I
work for (www.tractionplatform.com)
do html emails that has a video player in the email so when you open the email
in your email clients such as outlook the video streams straight into outlook
(email me if you want to see a campaign we ran for Audi  it rocks)



The only problem is the video just streams
into the html email, there are no player/pause/stop/volume controls in the
email.





Ill start a bounty on the wiki with
$50 if enough people this is of interest to them and other people can add to
it.









Cheers,



Dean

(personally Im happy to pop an
external player with a mp3 because I always have mine running when Im at
my pc but I can see why in companies this might be of interest).















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Tuesday, 7 November 2006
10:13 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Microsoft will enter VoIP market in earnest nextyear, says Ballmer





Unified messaging would
be nice. Not just having my VM's e-mailed to me, but to be able to manage them
from with Outlook (or any other mail client for that matter) would be nice. I
picture it sort of like an IMAP mailbox, and the mail client just has some kind
of functionality to recognize that the message is a VM and not a mail message
(so it could display length, date/time received, CID, and provide a
play button). 

Just my two cents.

Alex



On 11/7/06, Dean Collins [EMAIL PROTECTED] wrote:





http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm



There's
not much in the article so only click through if super interested but I'm
curious and looking for people's opinions.



What
application integration would you like to see between MS (either Office or
other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook
and number pop I'm kind of curious what other functionality there is to be
developed (I'd also like to see drop and drag from outlook into conference
calls.







What
would you like to see in asterisk, if we get some solid responses we'll see
about organizing some bounties to get it developed.









Regards, 

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial). 












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-- 
Alex Robar
[EMAIL PROTECTED] 








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[asterisk-users] Latest Debian and latest zaptel

2006-11-09 Thread Christian

Hi all,
Since i cant get latet beta of zaptel installed on the latest test version of 
Debian with kernel 2.6.17-2-686 can someone who is using debian give me some 
tips on how to get it working and installed?
Many thanks,
Christian


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Re: [asterisk-users] asterisk and norstar

2006-11-09 Thread Jorge Mendoza

Hi Gustavo,

Auto attendant is easy, voicemail I don't think so (there are not
extension information when call is back to Asterisk).
We use the following topology:
- pstn line - norstar (ext 123) - (fxo) asterisk

Jorge Mendoza

Gustavo Berman wrote:
 Hi there!

 We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a
 couple of m0x16. It has 5 external analog lines. It has no auto
 attendant, and no voicemail. So every incoming call is forwarded to a
 operator, she pick up the phone, talks to the caller and transfer the
 call to the right extension.

 We are in Argentina, so buying a star talk is out of the question,
 there is no selling of that in here.

 So, we want to use * as an auto attendant and voicemail for our 50
 extensions.

 Is there anybody who has done that?

 What topology do we have to use? :
 1) pstn line - (fxo) asterisk (fxs) - norstar
 or
 2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstar
 or
 3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk
 (fxo/2) - ata/2 - ( ext.321) norstar
 or
 4) pstnl line - norstar (ext. 123) - ata - (fxo) asterisk

 Any help please?
 I'm not a telephone systems specialist!

 Thanks!
 -- 
 Gustavo Berman
 Sysadmin
 Depto. Informatica
 Universidad Nacional del Comahue
 Centro Regional Universitario Bariloche
 

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Re: [asterisk-users] asterisk and norstar

2006-11-09 Thread Jorge Mendoza

Hi Gustavo,

I correct myself. Voicemail is possible if you make a supervised
transfer (I was talking about blind transfer).
Sorry for my too fast response.

Jorge Mendoza

===

Hi Gustavo,

Auto attendant is easy, voicemail I don't think so (there are not
extension information when call is back to Asterisk).
We use the following topology:
- pstn line - norstar (ext 123) - (fxo) asterisk

Jorge Mendoza

Gustavo Berman wrote:
 Hi there!

 We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a
 couple of m0x16. It has 5 external analog lines. It has no auto
 attendant, and no voicemail. So every incoming call is forwarded to a
 operator, she pick up the phone, talks to the caller and transfer the
 call to the right extension.

 We are in Argentina, so buying a star talk is out of the question,
 there is no selling of that in here.

 So, we want to use * as an auto attendant and voicemail for our 50
 extensions.

 Is there anybody who has done that?

 What topology do we have to use? :
 1) pstn line - (fxo) asterisk (fxs) - norstar
 or
 2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstar
 or
 3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk
 (fxo/2) - ata/2 - ( ext.321) norstar
 or
 4) pstnl line - norstar (ext. 123) - ata - (fxo) asterisk

 Any help please?
 I'm not a telephone systems specialist!

 Thanks!
 -- 
 Gustavo Berman
 Sysadmin
 Depto. Informatica
 Universidad Nacional del Comahue
 Centro Regional Universitario Bariloche
 

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[asterisk-users] Re: DTMF Corruption Problem

2006-11-09 Thread Justin Tunney


Sup pigs,

I think I found the solution to the problem!

It's a one line of code fix to rtp.c that people have been going daffy
about for almost a year!  A patch was uploaded in August but for some
reason it hasn't found its way in to 1.2.

Here's the bug:

http://bugs.digium.com/view.php?id=5970

Direct link to patch for the lazy:

http://bugs.digium.com/file_download.php?file_id=11337type=bug

On 11/8/06, Justin Tunney [EMAIL PROTECTED] wrote:

Asterisk People,

I'm currently using Asterisk and with a SIP voip provider and I'm
having problems where DTMF input in my IVR app is getting corrupted
intermittently.

For example, if someone enters 1025, it may come though correctly as
1025, or it may come trough as 10025, or 100255.  DTMF digits will
just double up.

This doesn't happen all the time.  Asterisk will just pick times to
not be very friendly with DTMF, and other times it will just work
flawlessly.

I'm using RFC2833 on:

Linux hostname 2.6.9-42.0.2.ELsmp #1 SMP Wed Aug 23 00:17:26 CDT 2006
i686 i686 i386 GNU/Linux

with Asterisk 1.2.13.

Also, I am not using a zaptel timer.  Could this possibly be causing
problems with DTMF??

Thanks!

--
Justin Tunney


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Re: [asterisk-users] porting numbers away from packet 8?




More than "1 incoming line" will depends on your 
provider. I have a SIP provider that will send me up to 10 channels at a time. I 
personally use myphonecompany.com. They have been really good for 
me.

And now for the disclaimer:No I do not work for 
them. Just a reall happy customer for origination,

Dovid

  - Original Message - 
  From: 
  Dean Collins 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, November 09, 2006 10:49 
  PM
  Subject: [asterisk-users] porting numbers 
  away from packet 8?
  
  
  Does anyone know if its possible 
  to port a number AWAY from packet8?
  
  Ive been with them for 2 years 
  and really want to move to an IAX based service so I can have more than 1 
  incoming line at a time.
  
  
  
  
  Cheers,
  
  Dean
  
  
  
  

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Re: [asterisk-users] Quick Q...


Post away.
- Original Message - 
From: Jay Moore [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, November 09, 2006 6:58 PM
Subject: [asterisk-users] Quick Q...


Before I make any serious gaffes, is this an acceptable place to post 
PHPAGI questions as well?  I can't seem to find a dedicated mailing list 
for it.  If not, any suggestions?


Thanks,
Jay
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Re: [asterisk-users] Voxee lag problems ?




I have the same issue. Just went in to my box. 
Seems I am still registering with them. This is from when I tested them. The 
call quality was horrible. If you want IAX specificly I would recomend 
teliax.com. Call quality is great and during business hours some one actually 
answers the phone.


  - Original Message - 
  From: 
  [EMAIL PROTECTED] 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, November 09, 2006 9:11 
  PM
  Subject: Re: [asterisk-users] Voxee lag 
  problems ?
  
  Hi Vicky,
  
  I used to use their termination services, but I had the same problems ... 
  It is impossible to work with that latency ... A lot of gaps and dead spots 
  ...
  
  If you know ... I'm looking for a good termination provider that I can 
  use the combination IAX/iLBC ... If you know some .. can you please tell me 
  ?
  
  Thanks,
  
  -- 
Original message -- From: Vicky [EMAIL PROTECTED] 
Anyone having problems with voxee since last few days or is it just me ? 
In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms 
latency . Most of time it is 20 ms or so but when i start sending traffic to 
them latency increases to 1000 ms or even LAGGED ( also shows high in 
peak time even when no high latency 
).Noproblemswithanyotherprovider . 
Anyone else having same problem ? 
  
  

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Re: [asterisk-users] Off-Site Extensions That Would Show As In-Use?




Are you trying to get FOP to monitor the SIP 
account that you are using to dial the cell phone on ?


  - Original Message - 
  From: 
  Alexander 
  Burke 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, November 08, 2006 11:07 
  PM
  Subject: [asterisk-users] Off-Site 
  Extensions That Would Show As In-Use?
  Hello, list!I'd 
  like to create an extension that points to an offsite location (a number on 
  the PSTN), the purpose of which would be to see if that offsite location is 
  still on a call forwarded to it by Asterisk. This way a receptionist could 
  choose to transfer calls to a mobile phone only if it's finished with the last 
  call the receptionist forwarded to it.If I configure a custom 
  extension with the destination SIP/TrunkName/NXXNXX, the calls transfer 
  fine but don't show as busy using the Flash Operator Panel (as an 
  example).Any thoughts?Thanks in advance,Alex-- 
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
  
  

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RE: [asterisk-users] porting numbers away from packet 8?

2006-11-09 Thread Dean Collins









Hi David,

Packet 8 are restricted to a single call
because they use an ATA which I then route into my asterisk server via a tdm400p







Cheers,



Dean















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Thursday, 9 November 2006
7:35 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
porting numbers away from packet 8?







More than 1 incoming line will depends on your
provider. I have a SIP provider that will send me up to 10 channels at a time.
I personally use myphonecompany.com. They have been really good for me.











And now for the disclaimer:
No I do not work for them. Just a reall happy customer for origination,











Dovid







- Original Message - 





From: Dean Collins 





To: asterisk-users@lists.digium.com 





Sent: Thursday, November
09, 2006 10:49 PM





Subject: [asterisk-users]
porting numbers away from packet 8?









Does anyone know if its possible to port a number
AWAY from packet8?



Ive been with them for 2 years and really want to
move to an IAX based service so I can have more than 1 incoming line at a time.









Cheers,



Dean











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RE: [asterisk-users] Modprobe Zaptel

2006-11-09 Thread Julian Varanini



Hi Eric,

Tried that but I am still getting the same error.

Thanks

Julian



 Date: Thu, 9 Nov 2006 17:25:02 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modprobe Zaptel  Julian Varanini wrote:  Hi,Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found"  You need to edit /usr/src/linux/Makefile to remove make the EXTRAVERSION  variable equal to "-12" rather than the "-12somethingsomethingsomething"  it is now. No need to recompile the kernel, just change the make file  and recompile and reinstall zaptel. ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Modprobe Zaptel

Then the make install in the Zaptel directory didn't work or installed 
it in the wrong location.


Julian Varanini wrote:

Hi Eric,
 
Tried that but I am still getting the same error.
 
Thanks
 
Julian





Date: Thu, 9 Nov 2006 17:25:02 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modprobe Zaptel  Julian 
Varanini wrote:  Hi,Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile 
and attempt a modprobe I get module zaptel not found  You need to edit /usr/src/linux/Makefile to remove make the EXTRAVERSION  variable equal to 
-12 rather than the -12somethingsomethingsomething  it is now. No need to recompile the kernel, just change the make file  and recompile and 
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RE: [asterisk-users] Modprobe Zaptel

2006-11-09 Thread Darryl Dunkin




After 
running 'make install', do a 'depmod -a'.

Then 
check /lib/modules for the file:
find 
/lib/modules | grep zaptel

Be sure 
the path/lib/modules/kernel/extra/zaptel.ko matches up with your 
currently running kernel (from uname-a) as that is where it will be 
checking.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Julian 
VaraniniSent: Thursday, November 09, 2006 15:21To: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Modprobe 
Zaptel

Hi,Can 
  someone walk me through compiling and loading the Zaptel 1.2.10 driver for 
  Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get 
  "module zaptel not 
found"ThanksJulian
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[asterisk-users] Powering SNOM 200 phones?

2006-11-09 Thread Brad Templeton


Ok, not exactly an Asterisk problem, but...

I picked up some SNOM 200 phones because SNOM's have been recommended for use
with Asterisk and they have line buttons that can subscribe to presence.

However, they don't appear to power up when connected to my Negear FS108P,
which is an 802.3af Power-over-ethernet capable hub.   I am pretty sure
these are the SNOM 200b, in that the ethernet connectors are at the
back rather than on the bottom, and there doesn't even seem to be
a jack for plugging in any other kind of power adapter (and I don't
have another one.)

Anybody had experience with these phones and powering them?  Is it
just an icompatability with the Netgear, or do I have 2 dead phones?
Would getting a different PoE box be a good idea?  (Frys has the
airlink for $29 from time to time, which is a great price.  Otherwise
many older PoE boxes tend to cost more than the modern cheaper phones
they might power.)


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[asterisk-users] DTMF problems with IVR - What DMTF Tx method

2006-11-09 Thread Bruce Ferrell



I'm having problems with a new asterisk PBX install.  the phones/ATAs 
are all linksys/cisco.  They all worked before with a commercial softswitch.


Most of the linksys devices offer auto, inband, INFO and AVT.   I'm 
looking for suggestions.


Thanks in advance

--
One day at a time, one second if that's what it takes

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Re: [asterisk-users] Powering SNOM 200 phones?

2006-11-09 Thread Christopher Aloi


Hey Brad -

I have a Snom 200 at the office, If I remember correctly the power is
about the size of an RJ11 jack; it's a weird connector.

I haven't used PoE with the 200 though.

Not sure if that helps or not :)

-chris

On 11/9/06, Brad Templeton [EMAIL PROTECTED] wrote:

 Ok, not exactly an Asterisk problem, but...

 I picked up some SNOM 200 phones because SNOM's have been recommended for use
 with Asterisk and they have line buttons that can subscribe to presence.

 However, they don't appear to power up when connected to my Negear FS108P,
 which is an 802.3af Power-over-ethernet capable hub.   I am pretty sure
 these are the SNOM 200b, in that the ethernet connectors are at the
 back rather than on the bottom, and there doesn't even seem to be
 a jack for plugging in any other kind of power adapter (and I don't
 have another one.)

 Anybody had experience with these phones and powering them?  Is it
 just an icompatability with the Netgear, or do I have 2 dead phones?
 Would getting a different PoE box be a good idea?  (Frys has the
 airlink for $29 from time to time, which is a great price.  Otherwise
 many older PoE boxes tend to cost more than the modern cheaper phones
 they might power.)


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--
--
Christopher T Aloi
--




--
--
Christopher T Aloi
--
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[asterisk-users] Ask users.conf

2006-11-09 Thread mrdlnf

Hi Alls,In Asterisk-1.4 there is new config file, users.conf, but i don't know how mechanism between users.conf and sip/iax.conf, usually i add new user in sip.conf, but when i try use asterisk-gui, it write to 
users.conf
 and when i type sip list peer on asterisk console, there is no user that i create with asterisk-gui. Please give me some explanation coz i am newbie..Thanks-- Regards,
mrdlnf


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Re: [asterisk-users] Off-Site Extensions That Would Show As In-Use?

2006-11-09 Thread Alexander Burke





Dovid B wrote:

  
  
  
  
  Are you trying to get FOP to monitor
the SIP account that you are using to dial the cell phone on ?

The SIP extension, yes. So, as long as a call that has been forwarded
to that cell phone is still in progress, that extension should still
show busy.

Thanks again,
Alex
-- 
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada


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Re: [asterisk-users] Powering SNOM 200 phones?

2006-11-09 Thread Paul Hales


I had a 200, and it worked fine with POE.

The standard power connector was the RJ-11 style as mentioned below.
Weird item that one.

The successor to the 200, known as a 190 does NOT support poe, while the
320 does.

later,

PaulH

On Thu, 2006-11-09 at 22:13 -0500, Christopher Aloi wrote:
  Hey Brad -
 
  I have a Snom 200 at the office, If I remember correctly the power is
  about the size of an RJ11 jack; it's a weird connector.
 
  I haven't used PoE with the 200 though.
 
  Not sure if that helps or not :)
 
  -chris
 
  On 11/9/06, Brad Templeton [EMAIL PROTECTED] wrote:
  
   Ok, not exactly an Asterisk problem, but...
  
   I picked up some SNOM 200 phones because SNOM's have been recommended for 
   use
   with Asterisk and they have line buttons that can subscribe to presence.
  
   However, they don't appear to power up when connected to my Negear FS108P,
   which is an 802.3af Power-over-ethernet capable hub.   I am pretty sure
   these are the SNOM 200b, in that the ethernet connectors are at the
   back rather than on the bottom, and there doesn't even seem to be
   a jack for plugging in any other kind of power adapter (and I don't
   have another one.)
  
   Anybody had experience with these phones and powering them?  Is it
   just an icompatability with the Netgear, or do I have 2 dead phones?
   Would getting a different PoE box be a good idea?  (Frys has the
   airlink for $29 from time to time, which is a great price.  Otherwise
   many older PoE boxes tend to cost more than the modern cheaper phones
   they might power.)
  
  
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  --
  --
  Christopher T Aloi
  --
 
 
 

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Re: [asterisk-users] Powering SNOM 200 phones?

2006-11-09 Thread Brad Templeton

On Fri, Nov 10, 2006 at 02:34:31PM +1100, Paul Hales wrote:
 
 I had a 200, and it worked fine with POE.
 
 The standard power connector was the RJ-11 style as mentioned below.
 Weird item that one.
 
 The successor to the 200, known as a 190 does NOT support poe, while the
 320 does.
 

Yeah, these have an extra unmarked rj-11 on the bottom next to two
covered holes (with nothing but pc board behind) where the ethernet
would be on the old model of snom 200 if I read the manual right.

So that's the power.   So I guess the only way to find out if
they just don't talk to my netgear POE (which does power
my grandstream 2000) is to find different POEs.  Or buy the power
supplies which don't seem to be very expensive -- or are there different
models of snom power supplies?   It is suggested the 190 takes 5v,
not 48v.
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Re: [asterisk-users] DTMF problems with IVR - What DMTF Tx method