Re: [Asterisk-Users] Re: delay in MeetMe

[amna saleem]

Hi!
I am using Asterisk-1.2.9.1
Zaptel 1.2.6
And my system has Linux Kernal 2.4
 
Best Regards,
Amna
 
On 6/14/06, Tony Mountifield <[EMAIL PROTECTED]> wrote:
In article <[EMAIL PROTECTED]
>,amna saleem <[EMAIL PROTECTED]> wrote:> Hi All!>> I am facing some delay in conferencing.> Using DIAX for Voip calls ,no hardware used yet
> I am using MeetMe to achieve conferencing  and am having a lot of delays.> Can anyone tell me how to reduce the delayWhat version of Zaptel are you using, what version of Asterisk, andwhich Linux kernel does your system have?
CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: 
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Re: [Asterisk-Users] delay in MeetMe

[amna saleem]

No , actually I am using Asterisk-1.2.9.1
I will try the q option though
 
Thanks and regards,
Amna  
On 6/14/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
I assume you are using 1.0.x.  Add the "q" option to the Meetmeextension.  1.0.x has a known issue where enter/exit sounds cause
increasing delays.amna saleem wrote:>   Hi All! I am facing some delay in conferencing.>> Using DIAX for Voip calls ,no hardware used yet>> I am using MeetMe to achieve conferencing  and am having a lot of delays.
>> Can anyone tell me how to reduce the delay Regards,>> Amna Saleem>>> 
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AW: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06

[Marc Rohlfing]

  Good morning,

> Why still use mpg123?
> Start using format_mp3 from asterisk-addons and your * will 
> play mp3 by itself...

good point - did that, and everything's working again. Thanks!

Marc Rohlfing

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Re: [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points

[John Joseph]

--- Markus Schuster <[EMAIL PROTECTED]> wrote:

> Could you please post some details (or even better:
> write them in some sort
> of Wiki) on the configuration you did on the Nokia?
> I'm thinking about buying a Nokia E60 but after a
> short web search there
> seem to be some problems about the correct
> configuration of the phone. 

  I tried to put some details on Voip-info.org ,
please check the link 
http://www.voip-info.org/wiki/view/Nokia
   thanks 
 Joseph John 




Send instant messages to your online friends http://uk.messenger.yahoo.com 
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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[Santosh Rao]

asterisk has a extremely cool documentation. The wiki has everything a newbie 
like me could hope for.. with samples and everyhting./. where as we are having 
a very dificult time finding proper documentation or samples and stuff like 
thtt for SER.. 
may be if someone good with SER could update ther voip-info/wiki and write some 
basics abt the ser.cfg or somethjing .. then it would be great. 

Regards
Santosh Rao


Martin Joseph wrote:
> >
>On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote:
>
>> If you do this, and not have Asterisk in the call setup path, your 
>> going to lose the ability to do a lot of features. What about 
>> black/white lists, rate centers, pic codes, intra company extension 
>> dialling and other advanced features?
>>
>> Sure, you might be able to do them with SER but good luck trying to 
>> find documentation.
>>
>So, your saying asterisk has better documentation?  I just want to be 
>sure I understand you   ;~)
>
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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[Santosh Rao]

asterisk has a extremely cool documentation. The wiki has everything a newbie 
like me could hope for.. with samples and everyhting./. where as we are having 
a very dificult time finding proper documentation or samples and stuff like 
thtt for SER.. 
may be if someone good with SER could update ther voip-info/wiki and write some 
basics abt the ser.cfg or somethjing .. then it would be great. 

Regards
Santosh Rao


Martin Joseph wrote:
> >
>On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote:
>
>> If you do this, and not have Asterisk in the call setup path, your 
>> going to lose the ability to do a lot of features. What about 
>> black/white lists, rate centers, pic codes, intra company extension 
>> dialling and other advanced features?
>>
>> Sure, you might be able to do them with SER but good luck trying to 
>> find documentation.
>>
>So, your saying asterisk has better documentation?  I just want to be 
>sure I understand you   ;~)
>
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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[Shaun Hofer]

www.onsip.org
One of the best places for ser info 

-Shaun

On Wednesday 14 June 2006 11:44, Kelvin Williams wrote:
> Has anyone ever published a concise howto or good documentation on how the
> two interrelate? and Configurations..
>
> On 6/13/06, BILL GITONGA <[EMAIL PROTECTED]> wrote:
> > Asterisk does to scale well. Use OpenSER or SER as a
> > front end to asterisk. Make all the sip traffic go
> > through ser and only go to Asterisk for voicemail, IVR
> > i.e media stuff. If you connect to the PSTN using sip,
> > then SER would be used for routing all PSTN calls.
> >
> > --- Erick Perez <[EMAIL PROTECTED]> wrote:
> > > While reading about how to maximize capabilities in
> > > asterisk i have
> > > read about SER and OpenSER.
> > >
> > > The sites do not explain to newbies (maybe that's on
> > > purpose) what are
> > > the benefits of using those products tied with
> > > asterisk (or is SER an
> > > asterisk replacement??)
> > >
> > > Can someone give me an idea of what's the usage for
> > > open(ser) and asterisk?
> > > is it for scalability?
> > > should I run it in the same box as asterisk or
> > > separated?
> > > does it add more functions to asterisk?
> > > or is the main function to better handle SIP over
> > > firewalls (due to
> > > SIP over TCP support)?
> > >
> > > Thanks for the explanation.
> > >
> > >
> > >
> > > --
> >
> > 
> >
> > > Erick Perez
> >
> > 
> >
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> >
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> >
> >
> >
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[Asterisk-Users] ISDN in Japan

[KokMeng Loh]

Hi,

Has anyone gotten the Billion BiPAC PCI ISDN card to work with the ISDN 
(INS net64) and Asterisk in Japan?


http://www.billion.com/product/isdn/bipacpciv30.htm

Regards,
KokMeng.

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Re: [Asterisk-Users] ztdummy

[Martin Joseph]

On Jun 13, 2006, at 7:52 PM, Josué Conti wrote:


   Doug,
If you it will not have hardware and if ztdummy will not have 
installed its moh will not function correctly


I believe this is no longer be true with the "new" Native music on 
hold...


Marty


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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[Martin Joseph]

On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote:

If you do this, and not have Asterisk in the call setup path, your 
going to lose the ability to do a lot of features. What about 
black/white lists, rate centers, pic codes, intra company extension 
dialling and other advanced features?


Sure, you might be able to do them with SER but good luck trying to 
find documentation.


So, your saying asterisk has better documentation?  I just want to be 
sure I understand you   ;~)


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[Asterisk-Users] Asterisk-1.0.9 Atxfer

[Josué Conti]

Hello All.
As I can edit the parameter featurestimeout in asterisk-1.0.9? Exists some option so that I can increase the time between digits of atxfer?featurestimeout = 500In the Asterisk version 1.0.9, features.conf does not consist this parameter, everything indicates that this version uses 500ms as value timeout standard between dialings, or either, is a very short time between dialings.
Please help me?
Best Regards
 
Josué
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RE: [Asterisk-Users] Will 200KB/s drive access be OK for voicemailstorage?

[Boris Bakchiev]

Its slow :) It will give you some delays but it will not be noticeable
(most voice files are 5-100kb, so it should be ok... But writing to
them.. Not sure.. It should be ok as well I'm guessing as kernel will
provide some caching (since you have G and not GS it has less ram, so
maybe chaching is not an option) 
Best would be to get WL-500G. It has a USB port so you can plug a USB
memory stick into it. It will be faster and will give you more storage
cheaper then SD.


> I'm using a Linksys WRT54G router and it works great but it only has
> 4MB of storage.  One of my only options is to modify the router to
> accept SD cards (eg. 512MB) but the access time is only around 200
> KB/s.
> 
> Will this be fast enough to store voicemails?   ...or do I need
> another (faster) storage device?
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RE: [Asterisk-Users] OPENSER / SER and Asterisk

[Douglas Garstang]

If you do this, and not have Asterisk in the call setup path, your going to 
lose the ability to do a lot of features. What about black/white lists, rate 
centers, pic codes, intra company extension dialling and other advanced 
features?
 
Sure, you might be able to do them with SER but good luck trying to find 
documentation.

-Original Message- 
From: BILL GITONGA [mailto:[EMAIL PROTECTED] 
Sent: Tue 6/13/2006 7:14 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk




Asterisk does to scale well. Use OpenSER or SER as a
front end to asterisk. Make all the sip traffic go
through ser and only go to Asterisk for voicemail, IVR
i.e media stuff. If you connect to the PSTN using sip,
then SER would be used for routing all PSTN calls.

--- Erick Perez <[EMAIL PROTECTED]> wrote:

> While reading about how to maximize capabilities in
> asterisk i have
> read about SER and OpenSER.
>
> The sites do not explain to newbies (maybe that's on
> purpose) what are
> the benefits of using those products tied with
> asterisk (or is SER an
> asterisk replacement??)
>
> Can someone give me an idea of what's the usage for
> open(ser) and asterisk?
> is it for scalability?
> should I run it in the same box as asterisk or
> separated?
> does it add more functions to asterisk?
> or is the main function to better handle SIP over
> firewalls (due to
> SIP over TCP support)?
>
> Thanks for the explanation.
>
>
>
> --
>

> Erick Perez
>

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>  
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>


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Re: [Asterisk-Users] Linksys SRW224P POE Switch

[Nicholas Kathmann]

The Cisco IP phones that end in 1, for instance 7941, 7961, 7971, all 
support 802.3af standards, and were built just for that.


Thanks,
Nick

Rich Adamson wrote:

Tom wrote:

At 05:24 AM 6/10/2006, you wrote:

" What will you be powering with it?  I bought one to power Cisco IP
phones but realized that it will not power them before it arrived."

What obscure cisco phones are you using ?


7960G  - pretty obscure huh?


You may only need a different fly lead if you have older pre 
standard phones

otherwise it should work ok.


Pay attention Fadge:  There are many types of POE and at least two 
types that use 48v and are somewhat 802.3af compliant.  Older POE 
like Cisco take their power over the unused pairs of an ethernet cable.


Most of the latest generation POE switches including the Linksys 
SRW224P provide their power on the data pairs, not the unused pairs.  
So if both the data and the power are on the same pairs, how do you 
make a cable adapter to work with the 7960G?


Its my understanding the 7960"G" model is .3af compliant, and the 
older 7960's are not. I think I read that on one of Cisco's pages, but 
not sure, and obviously I haven't tested it.


R.

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[Asterisk-Users] GXP-2000 Audio Quality

[Daniel Salama]

I have a client with about 16 GXP-2000. They complain that the audio  
quality is terrible after 2 or 3 simultaneous conversations. They are  
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u  
codec, I know they upstream bandwidth is the limiting factor and they  
most likely won't be able to have more than 3 simultaneous  
conversations, and if they're surfing the net and/or checking email,  
things will only get worse.


So, I purchased some g729 codec licenses and forced their sip peer  
configuration to g729 codec. We made sample test calls and were able  
to make 8 simultaneous calls. On the eighth call, the audio started  
to sound choppy. Then we dropped the eighth call and tested with 7.  
We could hear just fine on the GXP-2000 but the remote end heard us a  
bit choppy and/or with a robot-like voice. So, we kept dropping calls  
until they were of acceptable quality.


My question is, if they were using g729 which, in theory uses 8kbps  
plus overhead, they should have been just fine handling eight calls.  
All the computers were turned off on the network, so there shouldn't  
have been any other traffic but VoIP. Does anyone have any ideas?


How can I improve their audio quality? I requested BellSouth to  
upgrade their capacity but because of where they are located, the  
best they can get is 900Kbps/256Kbps, so the upstream continues to be  
the limiting factor.


I purchased a Dlink-1226G switch to allow me to control QoS on the  
LAN. I also upgraded their Netopia DSL router to the latest firmware  
which allows me to configure VLANs and DiffServ. All the computers  
are connected to the PC port on the phone because there is no  
available second wiring. Can anyone suggest how to configure the QoS  
settings on the phones, the Dlink and the Netopia?


While there was "no traffic" on the wire, pinging from/to the  
Asterisk box gave me about 47ms latency. When we went passed the 4th  
call, the latency started increasing significantly and when we got to  
8 calls, the latency was up in the 2000ms. Obviously, if anything I  
did in the QoS configuration gave VoIP a priority, then ICMP packets  
would have the lowest priority and I could understand that to be the  
reason for such result. However, I'm not sure I configured QoS  
properly and that's why I'm asking for help.


Thanks,
Daniel
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Re: [Asterisk-Users] how to hang the zap channel

[Kevin P. Fleming]

- Bartosz Wegrzyn - asterisk <[EMAIL PROTECTED]> wrote:

> When Voip users disconnect the 500 meeting is still active with the
> zap
> channel. How can I write extension to shut the channel down?

Make the VOIP users 'marked' users in MeetMe, and then also use the option to 
have the conference shut down when the last 'marked' user leaves.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] IAX2 Vs SIP cpu load

[Mike Fedyk]

Patrick wrote:

On Tue, 2006-06-13 at 23:47 +0800, Dinesh Nair wrote:
  

On 06/13/06 22:49 Colin Anderson said the following:


Although this may have changed in the newer 1.2.X series of Asterisk, I
believe that Asterisk does not support SMP from the perspective of
  
isnt asterisk multithreaded ? on a proper OS thread implementation, threads 
can migrate across CPUs, can't they ?



Afaik in 1.2.x IAX is single threaded. In 1.4 it is multithreaded.
In 1.2.x IAX uses two threads.  One to send, and one to receive.  In 1.4 
it will use more threads, but I don't know what the new threading model 
for IAX will be though.

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[Asterisk-Users] how to hang the zap channel

[Bartosz Wegrzyn - asterisk]

hello,

I got those extensions:

exten => 555,1,MeetMeCount(500|count)
exten => 555,2,Gotoif,$[${count} = 1]?6
exten => 555,3,Meetme,500|pMs|1234
exten => 555,4,Playback,goodbye
exten => 555,5,Hangup
exten => 555,6,Goto(from-internal-custom,556,1)
exten => 555,7,hangup

exten => 556,1,System(/bin/cp /etc/asterisk/1-test
/var/spool/asterisk/outgoing/)
exten => 556,2,Meetme,500|pMs|1234
exten => 556,3,Hangup

When users call 555 it automatically dials zap channel so
voip users canconference with PSTN users.

When Voip users disconnect the 500 meeting is still active with the zap
channel. How can I write extension to shut the channel down?

Thanks
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[Asterisk-Users] voip to voip bridge

[Erick Baum]

Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line?  Whenever we do this, the connected call is kinda lagged and the quality isn't always that great.  It seems to me this is just a problem with the inherent delay in the voip connections.  But I was wondering if there's any special configurations that could make the situation better?

 
Erick
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Re: [Asterisk-Users] Voiicemail NFS Cutting Out

[El Flynn]

BILL GITONGA wrote:


I have two asterisk systems that share voicemail on an
NFS. I recently upgraded to Asterisk 1.2.9.1. 
After the upgrade, the voicemail gets cut out after

about 5 seconds of recording. Any ideas on what might
be causing this?



What does it show on the CLI when this happens? More info would be helpful.


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Re: [Asterisk-Users] ztdummy

[Josué Conti]

   Doug,
If you it will not have hardware and if ztdummy will not have installed its moh will not function correctly
Regards
 
Josué 
2006/6/13, Doug Crompton <[EMAIL PROTECTED]>:
Well I don't use trunking or conferencing but I do use MOH and it worksfine as does VM and other things that would use timimg. Nothing I read is
very clear on this, just that you need it. Since I neglected to install itI am wondering what effects I should see.DougOn Tue, 13 Jun 2006, [ISO-8859-1] Josué Conti wrote:> Ztdummy is used when it does not have hardware's Digium installed in
> asterisk.  It also is used for MoH.> Best Regards> Josué>> 2006/6/13, Angelito Manansala <[EMAIL PROTECTED]>:> >
> > you only need that for conferencing ang trunking.> >*  Doug Crompton   **  Richboro, PA 18954  **  215-431-6307**  *
* [EMAIL PROTECTED]** http://www.crompton.com  *___
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Re: [Asterisk-Users] ztdummy

[Doug Crompton]

Well I don't use trunking or conferencing but I do use MOH and it works
fine as does VM and other things that would use timimg. Nothing I read is
very clear on this, just that you need it. Since I neglected to install it
I am wondering what effects I should see.

Doug

On Tue, 13 Jun 2006, [ISO-8859-1] Josu? Conti wrote:

> Ztdummy is used when it does not have hardware's Digium installed in
> asterisk.  It also is used for MoH.
> Best Regards
> Josu?
>
> 2006/6/13, Angelito Manansala <[EMAIL PROTECTED]>:
> >
> > you only need that for conferencing ang trunking.
> >


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [Asterisk-Users] ztdummy

[Josué Conti]

Ztdummy is used when it does not have hardware's Digium installed in asterisk.  It also is used for MoH.
Best RegardsJosué
2006/6/13, Angelito Manansala <[EMAIL PROTECTED]>:

you only need that for conferencing ang trunking.
On 6/14/06, Doug Crompton <[EMAIL PROTECTED]
> wrote: 
Question... is ztdummy required if you use no internal cards, in my casean SPA-3000 fxs/fxo ?  I have been running without it and see nothing not 
working. Using a 2.4 kernel, 1.2.9.1 *Doug*  Doug Crompton   *
*  Richboro, PA 18954  **  215-431-6307**  * * [EMAIL PROTECTED]
** http://www.crompton.com  *___ 
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Re: [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?

[Lachek Butalek]

While it's not the way I do it personally, I've been told the
absolutely easiest setup is [EMAIL PROTECTED] It is based on CentOS so
security updates should be coming down at a decent pace and I imagine
this could be set up to occur automatically. At that point, you pretty
much have a fire-and-forget Asterisk install with most of the bells
and whistles.

My personal choice is Slackware - combined with a 3rd party package
managing tool called Swaret you can set up a very slim, simple and
secure box in a very short amount of time and have do security patch
deployments automatically, much like CentOS/Redhat/etc. I'm personally
running Asterisk with FreePBX on an Apple PowerMac G3 266MHz using
Slackintosh, with great results so far.

Your system should be fine, though I'd probably dump some more RAM
into it if I were you - especially if you want to run [EMAIL PROTECTED], which 
uses
a MySQL database and the Apache webserver, among other things.

On 6/13/06, John Klimek <[EMAIL PROTECTED]> wrote:

First off, I'm sorry for sending so many messages to the list-serv.
Hopefully this will be my last for a while!

I was going to use my WRT54G router as a small Asterisk box, but I
forgot that I had a spare eMachines computer (Intel Celeron 633 MHz,
20GB HD, 64mb RAM).  Will this machine work OK for a very simple
dedicated home Asterisk box?

Also, what is easiest linux distribution to use and install?  All I
want is a simple Asterisk box that I can telnet into and have
voicemail, music-on-hold (MP3), etc...
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Re: [Asterisk-Users] ztdummy

[Angelito Manansala]

you only need that for conferencing ang trunking.On 6/14/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
Question... is ztdummy required if you use no internal cards, in my casean SPA-3000 fxs/fxo ?  I have been running without it and see nothing not
working. Using a 2.4 kernel, 1.2.9.1 *Doug*  Doug Crompton   **  Richboro, PA 18954  **  215-431-6307**  *
* [EMAIL PROTECTED]** http://www.crompton.com  *___
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[Asterisk-Users] ztdummy

[Doug Crompton]

Question... is ztdummy required if you use no internal cards, in my case
an SPA-3000 fxs/fxo ?  I have been running without it and see nothing not
working. Using a 2.4 kernel, 1.2.9.1 *

Doug


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?

[Mike Fedyk]

First, remove telnet from your vocabulary.  It should only be used over 
serial connections these days.  All other times, you should be using ssh.


Second, do you want the computer to be installed and running without any 
major software changes for a year or more?  Then use Centos or Ubuntu 
Dapper 6.06 or Debian Sarge 3.1.  Make sure you don't install the 
graphics as it can affect the latency of asterisk, especially on older 
hardware.


Third, I run asterisk on a PPro 200 at home, so your machine is beefy 
enough for sure.


And lastly, just give it a try, you'll learn a lot just making the effort.

Mike

John Klimek wrote:

First off, I'm sorry for sending so many messages to the list-serv.
Hopefully this will be my last for a while!

I was going to use my WRT54G router as a small Asterisk box, but I
forgot that I had a spare eMachines computer (Intel Celeron 633 MHz,
20GB HD, 64mb RAM).  Will this machine work OK for a very simple
dedicated home Asterisk box?

Also, what is easiest linux distribution to use and install?  All I
want is a simple Asterisk box that I can telnet into and have
voicemail, music-on-hold (MP3), etc...
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[Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?

[John Klimek]

First off, I'm sorry for sending so many messages to the list-serv.
Hopefully this will be my last for a while!

I was going to use my WRT54G router as a small Asterisk box, but I
forgot that I had a spare eMachines computer (Intel Celeron 633 MHz,
20GB HD, 64mb RAM).  Will this machine work OK for a very simple
dedicated home Asterisk box?

Also, what is easiest linux distribution to use and install?  All I
want is a simple Asterisk box that I can telnet into and have
voicemail, music-on-hold (MP3), etc...
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RE: [Asterisk-Users] spa3102 vs spa3000 differences?

[Damon Estep]

This was a topic covered a day or so ago.

I asked this same question, and my Cisco Voice product rep explained
that the 02 series has more memory to handle larger firmware images.

The two models take different firmware, and some newer features will not
be able to be implemented on the non 02 version once the image exceeds
the memory capacity.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Monday, June 12, 2006 7:23 AM
> To: Asterisk Users-List
> Subject: [Asterisk-Users] spa3102 vs spa3000 differences?
> 
> Anyone know what the differences are between the spa3000 and spa3102
> other then packaging?
> 
> R.
> 
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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[Kelvin Williams]

Has anyone ever published a concise howto or good documentation on how the two interrelate? and Configurations..  On 6/13/06, BILL GITONGA <
[EMAIL PROTECTED]> wrote:Asterisk does to scale well. Use OpenSER or SER as a
front end to asterisk. Make all the sip traffic gothrough ser and only go to Asterisk for voicemail, IVRi.e media stuff. If you connect to the PSTN using sip,then SER would be used for routing all PSTN calls.
--- Erick Perez <[EMAIL PROTECTED]> wrote:> While reading about how to maximize capabilities in> asterisk i have> read about SER and OpenSER.
>> The sites do not explain to newbies (maybe that's on> purpose) what are> the benefits of using those products tied with> asterisk (or is SER an> asterisk replacement??)>
> Can someone give me an idea of what's the usage for> open(ser) and asterisk?> is it for scalability?> should I run it in the same box as asterisk or> separated?> does it add more functions to asterisk?
> or is the main function to better handle SIP over> firewalls (due to> SIP over TCP support)?>> Thanks for the explanation. -->
> Erick Perez>> ___> --Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] OPENSER / SER and Asterisk

[BILL GITONGA]

Asterisk does to scale well. Use OpenSER or SER as a
front end to asterisk. Make all the sip traffic go
through ser and only go to Asterisk for voicemail, IVR
i.e media stuff. If you connect to the PSTN using sip,
then SER would be used for routing all PSTN calls.

--- Erick Perez <[EMAIL PROTECTED]> wrote:

> While reading about how to maximize capabilities in
> asterisk i have
> read about SER and OpenSER.
> 
> The sites do not explain to newbies (maybe that's on
> purpose) what are
> the benefits of using those products tied with
> asterisk (or is SER an
> asterisk replacement??)
> 
> Can someone give me an idea of what's the usage for
> open(ser) and asterisk?
> is it for scalability?
> should I run it in the same box as asterisk or
> separated?
> does it add more functions to asterisk?
> or is the main function to better handle SIP over
> firewalls (due to
> SIP over TCP support)?
> 
> Thanks for the explanation.
> 
> 
> 
> -- 
>

> Erick Perez
>

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Re: [Asterisk-Users] Linksys SRW224P POE Switch

[Cory Andrews]

There is an RJ45 cabling guide on the WIKI that shows how to create a 
reverse polarity crossover cable to power Cisco legacy PoE phones, and I can 
attest that it works with all the applications I have tried. 
Belkin/Powersense also makes an inline module for Cisco CDP that is 
relatively inexpensive.


Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message - 
From: "Mike Fedyk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 13, 2006 8:54 PM
Subject: Re: [Asterisk-Users] Linksys SRW224P POE Switch



Tom wrote:
Most of the latest generation POE switches including the Linksys SRW224P 
provide their power on the data pairs, not the unused pairs.  So if both 
the data and the power are on the same pairs, how do you make a cable 
adapter to work with the 7960G?

Maybe bridge the unused pairs with the data pairs?

I haven't tried it as I don't have any old style PoE, but it seems 
plausible.

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[Asterisk-Users] Will 200KB/s drive access be OK for voicemail storage?

[John Klimek]

I'm using a Linksys WRT54G router and it works great but it only has
4MB of storage.  One of my only options is to modify the router to
accept SD cards (eg. 512MB) but the access time is only around 200
KB/s.

Will this be fast enough to store voicemails?   ...or do I need
another (faster) storage device?
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Re: [Asterisk-Users] Linksys SRW224P POE Switch

[Rich Adamson]

Tom wrote:

At 05:24 AM 6/10/2006, you wrote:

" What will you be powering with it?  I bought one to power Cisco IP
phones but realized that it will not power them before it arrived."

What obscure cisco phones are you using ?


7960G  - pretty obscure huh?


You may only need a different fly lead if you have older pre standard 
phones

otherwise it should work ok.


Pay attention Fadge:  There are many types of POE and at least two types 
that use 48v and are somewhat 802.3af compliant.  Older POE like Cisco 
take their power over the unused pairs of an ethernet cable.


Most of the latest generation POE switches including the Linksys SRW224P 
provide their power on the data pairs, not the unused pairs.  So if both 
the data and the power are on the same pairs, how do you make a cable 
adapter to work with the 7960G?


Its my understanding the 7960"G" model is .3af compliant, and the older 
7960's are not. I think I read that on one of Cisco's pages, but not 
sure, and obviously I haven't tested it.


R.

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Re: [Asterisk-Users] delay in MeetMe

[Eric \"ManxPower\" Wieling]

I assume you are using 1.0.x.  Add the "q" option to the Meetme 
extension.  1.0.x has a known issue where enter/exit sounds cause 
increasing delays.


amna saleem wrote:

  Hi All!

 


I am facing some delay in conferencing.

Using DIAX for Voip calls ,no hardware used yet

I am using MeetMe to achieve conferencing  and am having a lot of delays.

Can anyone tell me how to reduce the delay

 


Regards,

Amna Saleem




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Now accepting new clients in Birmingham, Atlanta, Huntsville, 
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Re: [Asterisk-Users] Linksys SRW224P POE Switch

[Mike Fedyk]

Tom wrote:
Most of the latest generation POE switches including the Linksys 
SRW224P provide their power on the data pairs, not the unused pairs.  
So if both the data and the power are on the same pairs, how do you 
make a cable adapter to work with the 7960G?

Maybe bridge the unused pairs with the data pairs?

I haven't tried it as I don't have any old style PoE, but it seems 
plausible.

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RE: [Asterisk-Users] Linksys SRW224P POE Switch

[Tom]

At 05:24 AM 6/10/2006, you wrote:

" What will you be powering with it?  I bought one to power Cisco IP
phones but realized that it will not power them before it arrived."

What obscure cisco phones are you using ?


7960G  - pretty obscure huh?



You may only need a different fly lead if you have older pre standard phones
otherwise it should work ok.


Pay attention Fadge:  There are many types of POE and at least two 
types that use 48v and are somewhat 802.3af compliant.  Older POE 
like Cisco take their power over the unused pairs of an ethernet cable.


Most of the latest generation POE switches including the Linksys 
SRW224P provide their power on the data pairs, not the unused 
pairs.  So if both the data and the power are on the same pairs, how 
do you make a cable adapter to work with the 7960G?


Tom


Fadge


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: 10 June 2006 01:23
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Linksys SRW224P POE Switch

What will you be powering with it?  I bought one to power Cisco IP
phones but realized that it will not power them before it arrived.

It is still sitting sealed in the new packaging while I decide what
to do with it; otherwise I would tell you if it was loud or not.

Tom

At 02:04 PM 6/8/2006, Andres wrote:
>We are currently considering the Linksys POE switch for a small
>Asterisk office deployment.  There will be no separate wiring closet
>to put it in.  Can anybody tell me if this switch has a loud
>fan?  Users would not be able to tolerate a loud noise close
>by.  Otherwise we will go with a fanless desktop switch.
>
>Thanks,
>
>--
>Andres

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Re: [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?

[Mike Fedyk]

2.6.13 has the high precision timers changes.  And people have reported 
99.99 averages with that change.


I do not recommend compiling your own kernel unless you know what you 
are doing though.  If you just want to give it a try, then use Ubuntu 
6.06, Debian Etch, FC4 or FC5 (or some other distro that packages 2.6.13 
or higher).  If you want to use it in long term production, then just 
get a TDM410B card and use hardware timers.


Carl Youngblood wrote:

Thanks.  What is it in the 2.6.13-based kernel that improves timing?
Should I expect to see a significant improvement if I upgrade to it?

On 6/13/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:

IAX trunking and meetme conferences are some of the heaviest users of
zaptel timing.  I'd suggest if you don't have hardware timing (or at
least a 2.6.13 based kernel), then use SIP all the way or at least turn
off IAX trunking.

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[Asterisk-Users] message waiting

[Issac Simchayof]

Our message waiting stopped working on all of our polycom phones. I have
tested one of the phones on a different server and it work fine. The main
server that we just went into production with does not send the message
waiting info. There is no light or number of messages showing.

How can I trouble shoot this?

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RE: [Asterisk-Users] Hard drive write cache

[Boris Bakchiev]

> The cold hard truth is that if Asterisk cannot achieve 99.999% uptime
> without becoming much more expensive that a traditional PBX then it is
not
> a
> viable alternative.  Even elcheapo Key systems are rated for five
nines.
> That is what the telco world requires unless your just using Asterisk
in
> your basement as a hobby or as a one man company.

Well, you can pretty much guarantee 100% software uptime with asterisk.
The main causes of crashes of the working system are users.
If it works... don't touch it, do not logon to it... forget about it.

Create a minimalistic root system with busybox, have everything on CF on
IDE adapter, user UPS with shutdown to protect the CF (as they're prone
to failures on power loss) and you have yourself a VERY stable system.

You can use JFFS2 on block device to reduce the wear on CF but you will
not need it if you're not writing anything on CF (or have 2 CF's and md
them together.)

I have never seen PBX with guarantee of 99.999%. None of the
manufacturers will commit to that unless it is a highly redundant
system, but by then it's not elcheapo.

About fanless PC's.. A stock standard intel fan would lust longer then
you think unless its located in dusty and damp place.

I have a p3 that's been running for 5 years non-stop and it was still
going strong.. Half the capacitors started to leak on the motherboard
but the fan was still spinning. :) Now that's reliability!

> Redundant Servers is moving into the realm of non-competitive with
> Traditional PBX IMHO.
> 

More or less true. Any 100-200 extension highly redundant PBX system
will costs you more or less the same money.
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Re: [Asterisk-Users] Cisco 7960 BLA

[Lacy Moore - Aspendora]

Works great using SCCP.
On 6/13/06, Steve Glaus <[EMAIL PROTECTED]> wrote:
While I'm frantically scouring this list, does anyone have anyinformation about getting BLA (busy line appearance) working on Cisco 7960?
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Re: [Asterisk-Users] Audio cuts out

[Gary Richardson]

Hmm, out of curiosity, does anyone have experience with call recording and 3ware 9550SX cards? We're running a raid1 mirror.Our call recording load is more like 3-5 than 30-50, but I feel that there is some correlation between the two anyway at this point.
Thanks.On 6/12/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
We battled this same issue for a couple weeks, at about 30-50simultaneous recordings the audio would get all screwy.I looked at that solution but opted for something a little morepassive.  I use orkaudio to sniff rtp streams and mux them.  I have it
to perfect quality, the same as the monitor app in asterisk.  I think itis a much better solution than ramdisk since it is so passive and putsno strain or need for additional RAM on the asterisk machine itself.
Let the phone system be a phone system and not a recording device I say.Best part about the orkaudio project is Henri.  I had audio issues withorkaudio in the beginning but Henri re-worked his software to eliminate
my problems in a matter of days.  A true credit and great contributionto opensource software.Thanks,Steve TotaroGary Richardson wrote:> That could be an issue. Would recording to a ram drive solve the problem?
>> Thanks.>> On 6/12/06, *Steve Totaro* < [EMAIL PROTECTED]> [EMAIL PROTECTED]
>> wrote:>> Recording many simultaneous calls can cause this behavior too.>> Thanks,> Steve Totaro>>> 
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[Asterisk-Users] AGI and Video

[Rod Bacon]

I've been using Asterisk for over a year now, and think I've pretty much got 
it nailed form a voice perspective. We have just purchased a couple of Video 
phones to start experimenting with IP video, with a view to eventually 
building an IP media platform, such as Intel's HMP.

I have record/playback working in Voicemail, but am wondering what status 
Asterisk's AGI (specifically the record/playback fuctions) are at.

A couple of other things that I'd like to be able to do with video in the 
short term;

1. Build a video menu system to overlay an IVR (eg. press 1 for blah, press 2 
for something else, etc.)

2. Connect to an H.264 (or other codec) stream (eg. take a streaming feed from 
security camera, and attach it to an extension in the dialplan). I currently 
do this with audio, so I don't see video as being a huge extension to this.

Has anyone got any useful links or documentation on any of the above?

-- 
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600    Fax: +613 99401650
FWD: 512237                   ICQ: 5662270
==
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RE: [Asterisk-Users] Hard drive write cache

[shadowym]

 You still need a hard drive to run FreePBX as far as I know.  Either that
or burn out your 2GB CF drive after a couple months of constant writes.  I
would assume Apache and MySQL would need to go on the HD.  Voicemail can go
there as well.  Linux/Asterisk binaries and base config files can probably
go on the CF read only.  Would this work?  

Are there any how-to's around that have done something like this?  It is
starting to make a bit more sense to me to do it this way but I'm no expert.
If I did it this way I would not have a need for RAID so it will probably
come out to about the same price.


> -Original Message-
> From: Nicholas Kathmann 
> [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, June 13, 2006 1:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Hard drive write cache
> 
> They have IDE and even some SATA (not easily available) flash 
> drives also, some of which are over 80GB.  The more space you 
> get, the quicker it goes up in price with the larger models 
> costing more than most servers.  If you want, you can also 
> use an IDE to CF adapter. 
> 
> For the locking plugs, google  NEMA-L5 or NEMA-L6 and that 
> will show you what they look like.  They are readily 
> available in most hardware stores for low cost.  On the 
> server side, you can tie wrap the power cable to the rail or 
> something like that, but I would suggest just getting a 
> server that has the thumbscrew and clamp to hold the power 
> cord in place.  They are available on most IBM servers.  If 
> the power cord is mounted to the wall with staples or those 
> nail in C clamps, someone would have to go out of their way 
> to pull the power cord out. 
> 
> Another option if it's a really small installation is to use 
> a mini-itx fanless system.  We have 2 set up here (in a test 
> lab for now) with 1Ghz Via processors running up to ten 
> (that's all we've tried) concurrent calls with no problems.  
> These is no transcoding and echo cancellation running on 
> these.  Next it to try some Digium and Sangoma PCI cards in 
> them and see how they work.  I'll post the results when we 
> finish.  Such systems are extremely reliable as long as you 
> don't pull too much from the small power supplies, etc.  
> Regardless of what you use or what you do, trying to achieve 
> 5 nines reliability is going to require a whole rack full of 
> systems, storage, batteries, etc and a whole lot of 
> configuration and testing.  Even PBX systems (not all) 
> require downtime for firmware upgrades, etc.  Most people 
> don't bother since the systems aren't connected to data networks.
> 
> It was actually easier to pull the power cables out of some 
> of the PBX equipment (such as the Definity G3si) than it is 
> to pull the equipment out of the Cisco VoIP or IBM servers 
> combined with the right PDUs, etc.  
> 
> Thanks,
> Nick
> 
> shadowym wrote:
> > Thanks for the suggestions.
> >
> > CF is not an option for FreePBX which is a requirement for the 
> > installs I have in mind.  Astlinux on CF is a great option 
> otherwise.  
> > That is by far the simplest, cheapest, and suprisingly most 
> reliable 
> > solution I have come across so far.  If there was a half 
> decent (open 
> > source) GUI that could run on Astlinux on CF it would be a 
> no brainer IMHO.
> >
> > Physically locking down the server is not an option.  It 
> will be hung 
> > on the wall in place of where a traditional PBX would normally go.  
> > This is a telecom closet NOT a server rack environment.  
> UPS with auto 
> > shut down is just one link in the chain.  Do you have any further 
> > information of locking plugs?  I have not come across those 
> before.  
> > Of course in order for that to make sense I would need 
> locking plugs 
> > on both the server AND UPS end.  It has to be idiot proof.
> >
> > Think PBX and/or network appliance not computer server.  They are 
> > idiot proof so it is quite reasonable IMHO to expect the 
> same from an 
> > Asterisk server (or in my way of thinking, Asterisk network 
> appliance).
> >
> >   
> >> -Original Message-
> >> From: Nicholas Kathmann
> >> [mailto:[EMAIL PROTECTED]
> >> Sent: Tuesday, June 13, 2006 11:04 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [Asterisk-Users] Hard drive write cache
> >>
> >> If all you are worried about is the write cache on the 
> disks, why not 
> >> just put the system on a UPS set to shutdown the system in 
> the event 
> >> of power failure, then place both the UPS and asterisk 
> servers in a 
> >> locked rack.  In the event of a power failure (or someone knocking 
> >> the plug loose, which you can use locking plugs to further 
> mitigate), 
> >> the system will stay up on battery power then shut itself down to 
> >> prevent data corruption.  I doubt you will get that level 
> of uptime, 
> >> but there are other options to help achieve higher 
> reliability.  You 
> >> can run t

Re: [Asterisk-Users] Can this config sustain 30 users?

[Mike Fedyk]

http://www.voip-info.org/wiki/view/Asterisk+dimensioning
http://www.voip-info.org/wiki-Open+Source+VOIP+Software

Erick Perez wrote:

I appreciate all your help and posting.
I will then load (with test calls) using SIPP and astertest
will post back the result of this machine in question

any other open source stress test tool i can use?

Thanks,


On 6/13/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
>I'd suggest getting the slowest processor available new (At least 
1.5Ghz

>for AMD Athlon/Opteron and 2.xGhz  for Intel P4/Xeon)

I'm fond of underclocking. No heat problems for me, thank you.
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RE: [Asterisk-Users] Hard drive write cache

[shadowym]

Hi Andrew,

I think we are more on the same page than you think.  Fanless PC and CF are
exactly what I am shooting for.  However, as far as I know you need a hard
drive to run FreePBX.  Read only partitions make a lot of sense and I will
certainly pursue that further.  Not sure if I would need to put Voicemail on
a CF if I have a hard drive.  I could care less if someones voicemail
messages get corrupted during a power outage.  All I want is for the phone
system to come back up and work again.  GUARANTEED!

> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, June 13, 2006 1:12 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Hard drive write cache
> 
> On Tuesday 13 June 2006 15:36, shadowym wrote:
> > Physically locking down the server is not an option.  It 
> will be hung 
> > on the wall in place of where a traditional PBX would normally go.  
> > This is a telecom closet NOT a server rack environment.  
> UPS with auto 
> > shut down is just one link in the chain.  Do you have any further 
> > information of locking plugs?  I have not come across those 
> before.  
> > Of course in order for that to make sense I would need 
> locking plugs on both the server AND UPS end.
> > It has to be idiot proof.
> 
> Almost every telco closet I have been in (anything from 
> 5-person accounting firm to 750 person manufacturing facility 
> for Honda) has had a telco closet with a lock on the door.  
> Sometimes this closet was little more than a space under the 
> stairs to the basement and behind the water heater and 6 
> years of records, but the access was physically restricted.
> 
> Also, my Norstar MICS upstairs doesn't have locking plugs.  
> It's on a UPS that does not notify it of impending doom, and 
> it comes back up just fine most of the time[1].  My Linux 
> firewall is under the same constraints and also comes up 
> fine.  Why are you giving such heavy requirements for your 
> particular application?  What's wrong with a readonly / and 
> RAM drive /var and /tmp?  
> Store configs and voicemail on a flash drive or even a 
> journalled filesystem on a hard drive mounted synchronously.
> 
> [1] Ask any Norstar user how often Flash takes a big shit 
> when it loses power.  
> And there is no way to notify it.  I'd say at least 5-10% of 
> the time it has some problem, ranging from "stuck" voicemail 
> to corrupted voicemail to rare (but often enough) occurances 
> wherein you have to reinitialize the entire Flash voicemail system!
> 
> > Think PBX and/or network appliance not computer server.  They are 
> > idiot proof so it is quite reasonable IMHO to expect the 
> same from an 
> > Asterisk server (or in my way of thinking, Asterisk network 
> appliance).
> 
> I think you're aiming for too high a grade of idiot.  Any 
> standard fanless PC can achieve the kinds of uptimes you want 
> from your typical NEC Electra Elite or Norstar MICS without 
> resorting to the kind of circus act you're doing here.
> 
> -A.
> 
> 
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Re: [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?

[Carl Youngblood]

Thanks.  What is it in the 2.6.13-based kernel that improves timing?
Should I expect to see a significant improvement if I upgrade to it?

On 6/13/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:

IAX trunking and meetme conferences are some of the heaviest users of
zaptel timing.  I'd suggest if you don't have hardware timing (or at
least a 2.6.13 based kernel), then use SIP all the way or at least turn
off IAX trunking.

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Re: [Asterisk-Users] Queues and macros and agents

[Julian Lyndon-Smith]

Many thanks  :)

I was currently trying to add a macro to app_queue.c (like Dial) but 
will now abandon with indecent haste and use this !


Julian.

Kevin P. Fleming wrote:

- Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:

Now, I want to be able to use a device, rather than agents. So I can
use 
addQueueMember and add my SIP device. However, I still want to do a 
couple of things before the device is called.


This is what the Local channel (chan_local) is for.

If your SIP device is called "myfancyphone", then instead of adding 
SIP/myfancyphone to the queue using AddQueueMember, add (instead) Local/[EMAIL 
PROTECTED], and then in your dialplan:

[members]
exten => myfancyphone,1,...
exten => myfancyphone,n,...
exten => myfancyphone,n,Dial(SIP/${EXTEN})



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[Asterisk-Users] DISA Password Authenntication with Grandstream 488

[ITN Info - 11 - 30851536]

Hi

 

I can use now DISA settings
like this one when I set E1 card connected directly to Asterisk. In this way
every call dialed with pass 29 will be accepted. I have a billing which filters
caller ID number and address calls to each account with same caller ID number previously
set

 

[frommt] 

 

exten => 1536,1,Answer

exten => 1536,2,DigitTimeout(5)

exten => 1536,3,ResponseTimeout(10)

exten => 1536,4,Authenticate(29)

exten => 1536,5,DISA(no-password|brasil)

exten => 1536,6,Hangup

 

Now I need to add a
Grandstream 488 for DISA to remote landlines. So asterisk will receive phone
number from the landline connected to this grandstream and also the sip account
which is linked to Asterisk. But I cant decode caller phone number who dialed
to the landline connected to asterisk. Is that possible with Asterisk to create
a variable to collect a dialed password and then present that password which I
can read it and then manipulate that pass ? 

 

Regards from Brazil 

 

Kind
Regards,

 

 

 

Diretoria Comercial - Newton Medina

PABX    11.3085.1536

MSN [EMAIL PROTECTED] 

 

Rua
Augusta 2.212 SL 26 Jardins 01412001

São
Paulo - Brasil 

 

 






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[Asterisk-Users] OPENSER / SER and Asterisk

[Erick Perez]

While reading about how to maximize capabilities in asterisk i have
read about SER and OpenSER.

The sites do not explain to newbies (maybe that's on purpose) what are
the benefits of using those products tied with asterisk (or is SER an
asterisk replacement??)

Can someone give me an idea of what's the usage for open(ser) and asterisk?
is it for scalability?
should I run it in the same box as asterisk or separated?
does it add more functions to asterisk?
or is the main function to better handle SIP over firewalls (due to
SIP over TCP support)?

Thanks for the explanation.



--

Erick Perez

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[Asterisk-Users] RE: extensions.conf

[andrutto]

Hi

>Write a perl script that generates a mock 45,000 extensions.conf file, with 
>45,000 incrementing >extensions, throw in a couple of contexts. Start Asterisk 
>and see what happens.

I will do that first thing in the morning :)

>Actually i've done 50,000+ line dialplans using my Asterisk::LCR
>dialplan generator, and asterisk has been just fine with it.

Sound of relief :)

Thanks for help I will let you know about the results.

Cheers


--
Zobacz nowosci salonu moto w Madrycie >>> http://link.interia.pl/f1961

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Re: [Asterisk-Users] What is Echo?

[Zoa]

ECHOEcho ho.o.

Zoa

Mark Phillips wrote:

Actually this is an Elastic Impact. Throwing an object at another object
as suggested below could cause the kinetic energy possessed by the ball
to be diverted thus causing the ball to travel in a different direction
after impact.

This type of impact is commonly seen when insufficient kinetic energy is
presented to a much larger object thus causing the larger object to
dissipate the energy (usually as heat or sound) or in the case of most
hard surfaces to return the incoming energy along the same path it
arrived.

On Tue, 2006-06-13 at 13:44 -0400, C F wrote:
  

Echo is when you throw a basket ball on the floor and it bounces back,
the effect of the ball coming back to you is called Echo. If you go
into an empty big room and yell out I hate clinton you should hear the
walls agreeing with you and thats called echo.




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Re: [Asterisk-Users] SkypeOUT proxy

[Dovid Bender]

Get a VPS machine in the USOliver Vermeulen <[EMAIL PROTECTED]> wrote: Dose anybody know how to put skype behind a usa proxy ?Thanks,O___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Can this config sustain 30 users?

[Erick Perez]

I appreciate all your help and posting.
I will then load (with test calls) using SIPP and astertest
will post back the result of this machine in question

any other open source stress test tool i can use?

Thanks,


On 6/13/06, Colin Anderson <[EMAIL PROTECTED]> wrote:

>I'd suggest getting the slowest processor available new (At least 1.5Ghz
>for AMD Athlon/Opteron and 2.xGhz  for Intel P4/Xeon)

I'm fond of underclocking. No heat problems for me, thank you.
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--

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] Cisco 7960 BLA

[Steve Glaus]

While I'm frantically scouring this list, does anyone have any 
information about getting BLA (busy line appearance) working on Cisco 7960?


The last I heard was that this was  unsupported in Cisco's SIP firmware
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Re: [Asterisk-Users] What is Echo?

[Mark Phillips]

Actually this is an Elastic Impact. Throwing an object at another object
as suggested below could cause the kinetic energy possessed by the ball
to be diverted thus causing the ball to travel in a different direction
after impact.

This type of impact is commonly seen when insufficient kinetic energy is
presented to a much larger object thus causing the larger object to
dissipate the energy (usually as heat or sound) or in the case of most
hard surfaces to return the incoming energy along the same path it
arrived.

On Tue, 2006-06-13 at 13:44 -0400, C F wrote:
> Echo is when you throw a basket ball on the floor and it bounces back,
> the effect of the ball coming back to you is called Echo. If you go
> into an empty big room and yell out I hate clinton you should hear the
> walls agreeing with you and thats called echo.


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RE: [Asterisk-Users] Hard drive write cache

[Colin Anderson]

>[1] Ask any Norstar user how often Flash takes a big shit when it loses
power.  

lol...snap! aaahhh the memories. Makes the Asterisk burn seem minor by
comparision. 
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RE: [Asterisk-Users] Cell gateway for T-Mobile US??

[Mark Phillips]

T-Mobile do GSM, GPRS and EDGE and not GSM only as stated below.

Devices connected to their network typicly use GSM but may use GPRS if a
data plan is subscribed. Edge is available o those that have both an
Edge device and a data plan.

Not that I'm a T-Mo reseller or anything ;-}

On Mon, 2006-06-12 at 09:24 -0400, Brian C. Fertig wrote:
> Typically yes, as long as you can get power for them compatible with
> ours.  
> Tmobile is GSM.  Well only GSM.  They don't do anything else.  You can
> check
> the WIKI I have found a few smaller ones that will probably work but
> don't 
> remember what they are except that I found them there.
> 
> _.._
> Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
> Data/Telecom Engineer
> IT Administrator
> Planet Telecom, Inc
> Tampa, FL Office
> o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 
> SIP URI: [EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz,
> Steven
> Sent: Monday, June 12, 2006 9:03 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Cell gateway for T-Mobile US??
> 
> Most gateways I have found are only sold overseas.
> Do these work in the US?
> 
> My provider is T-Mobile (using their Blackberries).
> They support:
> GSM (I am pretty sure)
> GPRS
> EDGE
> 
> We get unlimited Cell to Cell minutes and would like to leverage the
> possible savings.
> 
> Does anyone know of a product that they have been happy with?
> 
> SIP or Analog is fine although SIP (or IAX) is preferred for the
> asterisk side.
> 
> Thanks.
>  
> Steven 
>  
> 
> 
> 
> Thank You,
> 
> Steven BerkHolz
> - MCSA - MCSE -
> Manager of Information Systems
> TESCO Group Companies
> Fax. 248-836-5101
> www.TESCOGroup.com
> 
> Board member of
> www.glimasoutheast.org
> 
> 
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> 
> This email was scanned by:  Mcafee GroupShield
>  CONFIDENTIAL DISCLAMER 
> All information provided in this email is considered confidential
> and proprietary of Planet Telecom, Inc. and Telecenter Inc.
> Use of this information by anyone other than the recipient or 
> sender will be considered in breach of agreement.
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Re: [Asterisk-Users] extensions.conf

[trixter aka Bret McDanel]

On Wed, 2006-06-14 at 00:50 +0400, Jean-Michel Hiver wrote:
> Actually i've done 50,000+ line dialplans using my Asterisk::LCR 
> dialplan generator, and asterisk has been just fine with it.

I have you beat, I have done over 500k when loading my country list that
I no longer maintain which is now at http://www.astbill.com.  It worked
although reloads and initial loads are noticably slow, and asterisk
seemed to be memory hungry.  I no longer do it that way and have had
better luck with it.

There are people that load a per user set of contexts out of a database,
so that they only have what is being used not everything known to man,
they have even put a expiry on the dialplan entries so they are freed
from memory if they arent used in X minutes.  I do not know if these
patches are public or not, but I do know they exist, this was for
someone who easily exceeded 2M lines and it was unmanagable otherwise,
given they have many boxes which all need that info and the way asterisk
processes dialplans, which becomes more evident under load.



-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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Re: [Asterisk-Users] Polycom Queues

[BJ Weschke]

On 6/13/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

Has anyone integrated Asterisk Queues with Polycom phones?

What I'd like to do is display the agent status next to their appearance. I 
don't see much discussion about this.
This is not the same thing as setting 1 against the appearance in the 
phone directory.



Yes. We've done it and it is at the following SVN branch:

http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/

What's still on the to-do list though is understand what messages the
Polycom phone needs to receive when an agent's state has changed as a
result of an action that didn't originate from the phone itself. That
hasn't proven to be a huge issue with the clients that are using it
already by us, but it's still something we should and intend to do to
"finish" the functionality.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Snom high SIP ping time

[Steve Glaus]

Mike Fedyk wrote:

Steve Glaus wrote:

Mike Hammett wrote:
(ICMP) pings were under 1 ms.  No amount of different Asterisk 
versions or phone firmware revisions seems to solve this.  All was 
well, then (as far as we know) without changes, it crapped out.
 
Any ideas?
  
I'm having much the same issues only I'm using Cisco 7960 phones. 
When I do a 'sip show peers' I'm getting times in excess of 300ms. A 
soft phone on the same network (x-lite), is reporting times of 4 ms. 
Related to this (I think), I'm getting audio issues. The person being 
called can hear the caller fine but the callee's voice drops in and 
out excessively.


I have qualify set to yes in the sip definitions for all the clients 
(Including the soft phone). Does anyone know what is causing this. 
I'm not aware what the sip ping times were earlier, but the audio 
issues seemed to have started spontaneously.

Do you have any problems when there are a low number of concurrent calls?
Do you ever get any messages saying the phones are unreachable?  Or 
lagged?

What kind of Internet connection do you have?
Do you have any problems with calls between phones on the same network 
(no routers in between)?

What model of switches do you have?
What model of Internet router do you have?
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Do you have any problems when there are a low number of concurrent calls?

Yes, from what I've seen and tested, it doesn't make a difference how 
many calls there are.



Do you ever get any messages saying the phones are unreachable?  Or lagged?

Yes, a phone not being used often enough will often report 
'UNREACHABLE'  When a call is made that changes



What kind of Internet connection do you have?

We have a synchronous T1 connection. The phones run on the same network 
as data.


Do you have any problems with calls between phones on the same network 
(no routers in between)?


There are some static issues but nothing like outside calling were the 
person being called drops off for 10 seconds at a time.


Switches used are 3com Baselines 2024

What model of Internet router do you have?

Cisco 1700 router


My main concern is the fact that the soft-phone is reporting only 4ms of 
delay and the cisco phones are reporting > 200. All the time, regardless 
of network bandwidth.



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Re: [Asterisk-Users] extensions.conf

[Jean-Michel Hiver]

Douglas Garstang a écrit :


-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extensions.conf


No limit in code imposed. Not sure about performance penalty for a
file that big, have you considered using ARA (Asterisk Realtime
Architecture)?

On 13 Jun 2006 21:06:52 +0200, andrutto <[EMAIL PROTECTED]> wrote:
   


Hi

Does anyone know how big extensions.conf can be?
I am trying to set up Asterisk which will have about 45 000 
 

lines in extensions.conf. Is there any limitation about the 
amount of lines in that file?
   



Write a perl script that generates a mock 45,000 extensions.conf file, with 
45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk 
and see what happens.
 

Actually i've done 50,000+ line dialplans using my Asterisk::LCR 
dialplan generator, and asterisk has been just fine with it.


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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[Asterisk-Users] Polycom Queues

[Douglas Garstang]

Has anyone integrated Asterisk Queues with Polycom phones?

What I'd like to do is display the agent status next to their appearance. I 
don't see much discussion about this.
This is not the same thing as setting 1 against the appearance in the 
phone directory.

Thanks
Doug.

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RE: [Asterisk-Users] extensions.conf

[Douglas Garstang]

> -Original Message-
> From: Moises Silva [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, June 13, 2006 1:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extensions.conf
> 
> 
> No limit in code imposed. Not sure about performance penalty for a
> file that big, have you considered using ARA (Asterisk Realtime
> Architecture)?
> 
> On 13 Jun 2006 21:06:52 +0200, andrutto <[EMAIL PROTECTED]> wrote:
> >
> > Hi
> >
> > Does anyone know how big extensions.conf can be?
> > I am trying to set up Asterisk which will have about 45 000 
> lines in extensions.conf. Is there any limitation about the 
> amount of lines in that file?

Write a perl script that generates a mock 45,000 extensions.conf file, with 
45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk 
and see what happens.

Doug.
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Re: [Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?

[Mike Fedyk]

Carl Youngblood wrote:

Our asterisk system gains access to the PSTN through a voip provider.
We have no digium or other telephony hardware in our system.  Do the
zttest results still matter to us?  Our results were as follows:

--- Results after 1007 passes ---
Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763
IAX trunking and meetme conferences are some of the heaviest users of 
zaptel timing.  I'd suggest if you don't have hardware timing (or at 
least a 2.6.13 based kernel), then use SIP all the way or at least turn 
off IAX trunking.

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Re: [Asterisk-Users] Snom high SIP ping time

[Mike Fedyk]

Steve Glaus wrote:

Mike Hammett wrote:
(ICMP) pings were under 1 ms.  No amount of different Asterisk 
versions or phone firmware revisions seems to solve this.  All was 
well, then (as far as we know) without changes, it crapped out.
 
Any ideas?
  
I'm having much the same issues only I'm using Cisco 7960 phones. When 
I do a 'sip show peers' I'm getting times in excess of 300ms. A soft 
phone on the same network (x-lite), is reporting times of 4 ms. 
Related to this (I think), I'm getting audio issues. The person being 
called can hear the caller fine but the callee's voice drops in and 
out excessively.


I have qualify set to yes in the sip definitions for all the clients 
(Including the soft phone). Does anyone know what is causing this. I'm 
not aware what the sip ping times were earlier, but the audio issues 
seemed to have started spontaneously.

Do you have any problems when there are a low number of concurrent calls?
Do you ever get any messages saying the phones are unreachable?  Or lagged?
What kind of Internet connection do you have?
Do you have any problems with calls between phones on the same network 
(no routers in between)?

What model of switches do you have?
What model of Internet router do you have?
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Re: [Asterisk-Users] Zork and Asterisk

[Simon P. Ditner]

A demo server is up and running the 0.2 beta for anyone who'd like to try
it out. Some brief instructions, and call in methods:

  http://uc.org/read/ZoIP%20Demo

Cheers,
spd

On Tue, 6 Jun 2006, John Todd wrote:

>
> http://www.boingboing.net/2006/06/05/play_zork_by_phone.html
>
> Let me preface this idea with one comment: I don't have the time to
> do this - I don't even have time to eat these days.  But someone out
> there has the cycles to do this... and it would be very cool.
>
> OK, so now Zork is attached to Asterisk, but using the
> less-than-clear Festival engine.  There are beta tests of the
> LumenVox speech recognition engine out there which tie directly into
> Asterisk.  Allison Smith (the voice of Asterisk) would almost
> CERTAINLY do a great dramatic reading of all of the text blocks
> within Zork.  I see an excellent opportunity for a demo server on
> some CLEC who would love to get some $ by opening up a few DIDs to a
> huge recip comp traffic load.  Even if it's just available via IAX2
> or SIP, this would be one of those "legends of the Net" in the next
> few years...
>
> JT
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[Asterisk-Users] [REPOST] Asterisk Realtime and "Ex-Girlfriend"

[Michael E. Kromer]

Hello all,

Last night I have successfully setup Asterisk Realtime with mysql. but I
have one problem regarding the "Ex-Girlfrind"-Functionality.

The example: I have a phone running on a specific extension (300) and I
want that one to call out via ZAP, but it simply gets IGNORED.

I have tried _X./300 => Dial(zap/g1/${EXTEN})

but what happens now is that (because of includes defined) all other
calls (_X.) are using this extension, even if somebody completely
different (for example 55) wants to call outside.

ZAP is definitly not the problem.

Help in this manner would be great. Thanks in advance.

-- 
Mit freundlichen Grüßen,
Best regards,

Michael E. Kromer
IT Specialist
Linux Professional Institute Certified (LPIC)

+--+
|   CC  Computer  Consultants  GmbH|
| ENTERPRISE.  IT.  BUSINESS.  |
|==|
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| Citrix Access Alliance Partner   |
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Re: [Asterisk-Users] Hard drive write cache

[Nicholas Kathmann]

They have IDE and even some SATA (not easily available) flash drives 
also, some of which are over 80GB.  The more space you get, the quicker 
it goes up in price with the larger models costing more than most 
servers.  If you want, you can also use an IDE to CF adapter. 

For the locking plugs, google  NEMA-L5 or NEMA-L6 and that will show you 
what they look like.  They are readily available in most hardware stores 
for low cost.  On the server side, you can tie wrap the power cable to 
the rail or something like that, but I would suggest just getting a 
server that has the thumbscrew and clamp to hold the power cord in 
place.  They are available on most IBM servers.  If the power cord is 
mounted to the wall with staples or those nail in C clamps, someone 
would have to go out of their way to pull the power cord out. 

Another option if it's a really small installation is to use a mini-itx 
fanless system.  We have 2 set up here (in a test lab for now) with 1Ghz 
Via processors running up to ten (that's all we've tried) concurrent 
calls with no problems.  These is no transcoding and echo cancellation 
running on these.  Next it to try some Digium and Sangoma PCI cards in 
them and see how they work.  I'll post the results when we finish.  Such 
systems are extremely reliable as long as you don't pull too much from 
the small power supplies, etc.  Regardless of what you use or what you 
do, trying to achieve 5 nines reliability is going to require a whole 
rack full of systems, storage, batteries, etc and a whole lot of 
configuration and testing.  Even PBX systems (not all) require downtime 
for firmware upgrades, etc.  Most people don't bother since the systems 
aren't connected to data networks.


It was actually easier to pull the power cables out of some of the PBX 
equipment (such as the Definity G3si) than it is to pull the equipment 
out of the Cisco VoIP or IBM servers combined with the right PDUs, etc.  


Thanks,
Nick

shadowym wrote:

Thanks for the suggestions.

CF is not an option for FreePBX which is a requirement for the installs I
have in mind.  Astlinux on CF is a great option otherwise.  That is by far
the simplest, cheapest, and suprisingly most reliable solution I have come
across so far.  If there was a half decent (open source) GUI that could run
on Astlinux on CF it would be a no brainer IMHO.

Physically locking down the server is not an option.  It will be hung on the
wall in place of where a traditional PBX would normally go.  This is a
telecom closet NOT a server rack environment.  UPS with auto shut down is
just one link in the chain.  Do you have any further information of locking
plugs?  I have not come across those before.  Of course in order for that to
make sense I would need locking plugs on both the server AND UPS end.  It
has to be idiot proof.

Think PBX and/or network appliance not computer server.  They are idiot
proof so it is quite reasonable IMHO to expect the same from an Asterisk
server (or in my way of thinking, Asterisk network appliance).

  

-Original Message-
From: Nicholas Kathmann 
[mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 13, 2006 11:04 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Hard drive write cache

If all you are worried about is the write cache on the disks, 
why not just put the system on a UPS set to shutdown the 
system in the event of power failure, then place both the UPS 
and asterisk servers in a locked rack.  In the event of a 
power failure (or someone knocking the plug loose, which you 
can use locking plugs to further mitigate), the system will 
stay up on battery power then shut itself down to prevent 
data corruption.  I doubt you will get that level of uptime, 
but there are other options to help achieve higher 
reliability.  You can run the OS and asterisk on a solid 
state disk, and have voicemail and whatever else you want to 
go to rotating disks.  That will also help with power usage 
on the server when using the UPS.  Industrial flash disks are 
said to have (but they really can't promise this) a 3 million 
hour MTBF.


Thanks,
Nick

shadowym wrote:

The cold hard truth is that if Asterisk cannot achieve 
  
99.999% uptime 

without becoming much more expensive that a traditional PBX 
  
then it is 

not a viable alternative.  Even elcheapo Key systems are 
  

rated for five nines.

That is what the telco world requires unless your just 
  
using Asterisk 


in your basement as a hobby or as a one man company.

Redundant Servers is moving into the realm of non-competitive with 
Traditional PBX IMHO.


I don't care about corruption of the CDR or any of the 
logging/database information.  All I care about is the ability make 
phone calls after power failure.  That IS the MAIN function 
  
of a PBX.  

Not call centers, databases, CDR, click 2 call, and all the 
  

other bells and whistles.

 

  
  

-O

[Asterisk-Users] MOH & Vegastream

[Issac Simchayof]

When ever we get a call through our VegaStream 50 FXO the MOH for that call
gets turned off. Anyway to troubleshoot this by looking at the log below?


Thanks

Issac


Jun 13 16:00:24 DEBUG[2443] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 2: Match Found
Jun 13 16:00:25 DEBUG[3107] rtp.c: Ooh, format changed from unknown to ulaw
Jun 13 16:00:26 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:29 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:34 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:36 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:39 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:43 DEBUG[2474] chan_iax2.c: Allocate call number
Jun 13 16:00:43 DEBUG[2474] chan_iax2.c: Registration created on call 1
Jun 13 16:00:43 NOTICE[2474] chan_iax2.c: Registration of 'issacs' rejected:
'Registration Refused' from: '207.174.202.4'
Jun 13 16:00:44 DEBUG[2474] chan_iax2.c: Allocate call number
Jun 13 16:00:44 DEBUG[2474] chan_iax2.c: Registration created on call 2
Jun 13 16:00:44 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:46 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:49 DEBUG[2443] chan_sip.c: Allocating new SIP dialog for
[EMAIL PROTECTED] - REGISTER (No RTP)
Jun 13 16:00:49 DEBUG[3113] app_queue.c: Device 'SIP/401' changed to state
'1' (Not in use) but we don't care because they're not a member of any
queue.
Jun 13 16:00:49 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:53 DEBUG[2443] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - NOTIFY (No RTP)
Jun 13 16:00:53 DEBUG[2443] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jun 13 16:00:54 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:56 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:59 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:04 DEBUG[2443] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jun 13 16:01:04 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:06 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:01:09 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes  
Jun 13 16:01:14 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:16 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:01:19 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:20 VERBOSE[2443] logger.c: -- Started music on hold, class
'default', on SIP/Vega-2127681168-4afe
Jun 13 16:01:20 DEBUG[2443] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 3: Match Found
Jun 13 16:01:24 DEBUG[3107] rtp.c: Got RTCP report of 56 bytes
Jun 13 16:01:28 VERBOSE[2443] logger.c: -- Stopped music on hold on
SIP/Vega-2127681168-4afe
Jun 13 16:01:28 DEBUG[2443] channel.c: Set channel SIP/Vega-2127681168-4afe
to write format ulaw


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Re: [Asterisk-Users] Hard drive write cache

[Andrew Kohlsmith]

On Tuesday 13 June 2006 15:36, shadowym wrote:
> Physically locking down the server is not an option.  It will be hung on
> the wall in place of where a traditional PBX would normally go.  This is a
> telecom closet NOT a server rack environment.  UPS with auto shut down is
> just one link in the chain.  Do you have any further information of locking
> plugs?  I have not come across those before.  Of course in order for that
> to make sense I would need locking plugs on both the server AND UPS end. 
> It has to be idiot proof.

Almost every telco closet I have been in (anything from 5-person accounting 
firm to 750 person manufacturing facility for Honda) has had a telco closet 
with a lock on the door.  Sometimes this closet was little more than a space 
under the stairs to the basement and behind the water heater and 6 years of 
records, but the access was physically restricted.

Also, my Norstar MICS upstairs doesn't have locking plugs.  It's on a UPS that 
does not notify it of impending doom, and it comes back up just fine most of 
the time[1].  My Linux firewall is under the same constraints and also comes 
up fine.  Why are you giving such heavy requirements for your particular 
application?  What's wrong with a readonly / and RAM drive /var and /tmp?  
Store configs and voicemail on a flash drive or even a journalled filesystem 
on a hard drive mounted synchronously.

[1] Ask any Norstar user how often Flash takes a big shit when it loses power.  
And there is no way to notify it.  I'd say at least 5-10% of the time it has 
some problem, ranging from "stuck" voicemail to corrupted voicemail to rare 
(but often enough) occurances wherein you have to reinitialize the entire 
Flash voicemail system!

> Think PBX and/or network appliance not computer server.  They are idiot
> proof so it is quite reasonable IMHO to expect the same from an Asterisk
> server (or in my way of thinking, Asterisk network appliance).

I think you're aiming for too high a grade of idiot.  Any standard fanless PC 
can achieve the kinds of uptimes you want from your typical NEC Electra Elite 
or Norstar MICS without resorting to the kind of circus act you're doing 
here.

-A.
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[Asterisk-Users] Are zttest results relevant on a system with no telephony hardware?

[Carl Youngblood]

Our asterisk system gains access to the PSTN through a voip provider.
We have no digium or other telephony hardware in our system.  Do the
zttest results still matter to us?  Our results were as follows:

--- Results after 1007 passes ---
Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763
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RE: [Asterisk-Users] Can this config sustain 30 users?

[Colin Anderson]

>I'd suggest getting the slowest processor available new (At least 1.5Ghz 
>for AMD Athlon/Opteron and 2.xGhz  for Intel P4/Xeon) 

I'm fond of underclocking. No heat problems for me, thank you. 
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Re: [Asterisk-Users] Snom high SIP ping time

[Steve Glaus]

Steven Ringwald wrote:

Steve Glaus wrote:

Mike Hammett wrote:
I don't know everything that's going on as someone else has been 
working on the project, but it hasn't really been going anywhere, so 
I had some questions.
 
We've got some Snom 320s with Asterisk 1.2.9.1 (I believe).  All was 
well (with a previous release), but the phones started to get real 
choppy.  We are also running a softphone at this location and it was 
fine.  The SIP qualify was returning ping times anywhere from 20 to 
70 ms over a sparsely used LAN.  Command prompt (ICMP) pings were 
under 1 ms.  No amount of different Asterisk versions or phone 
firmware revisions seems to solve this.  All was well, then (as far 
as we know) without changes, it crapped out.
  
I'm having much the same issues only I'm using Cisco 7960 phones. 
When I do a 'sip show peers' I'm getting times in excess of 300ms. A 
soft phone on the same network (x-lite), is reporting times of 4 ms. 
Related to this (I think), I'm getting audio issues. The person being 
called can hear the caller fine but the callee's voice drops in and 
out excessively.


I have qualify set to yes in the sip definitions for all the clients 
(Including the soft phone). Does anyone know what is causing this. 
I'm not aware what the sip ping times were earlier, but the audio 
issues seemed to have started spontaneously.


Anyone  have any idea regarding this? 



What codecs are you using? I have noticed that g729, for some reason, 
adds a lot of latency to the phone. Running on uLaw, however, I get 
times from sip show peers of around 5-14ms.


Steve

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I'm using ulaw all the way.
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Re: [Asterisk-Users] Can this config sustain 30 users?

[Mike Fedyk]

Erick Perez wrote:

I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.

24 SIP users and one E1 card in an Intel 945board (533 Front side bus)
with  1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB
533MHZ) and two 80GB SATA disks.
Can the box sustain the load? I can add another 1gb of ram if necessary.

Just PBX and voicemail, no fancy sutff like call recording...
maybe a simple autoattendant like "thank you for calling, please press
one for"
Let's just say that you shouldn't have trouble with four E1 lines in 
this system (as long as you have a hardware echo canceler).  Even with 
software echo can, this hardware is overkill.


I'd suggest getting the slowest processor available new (At least 1.5Ghz 
for AMD Athlon/Opteron and 2.xGhz  for Intel P4/Xeon) and get redundant 
power supplies, ECC memory, RAID 1 or higher and a nice UPS.


You may also think of looking at used previous generation high-end 
equipment.  It still has all of the good engineering and redundancy, it 
just isn't new and in many cases you don't need the latest speedy 
equipment to handle the load.  In many instances, the same money spent 
buying used high quality equipment will give you more reliability than 
buying cheap new equipment.

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[Asterisk-Users] Re: delay in MeetMe

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
amna saleem <[EMAIL PROTECTED]> wrote:
> Hi All!
> 
> I am facing some delay in conferencing.
> Using DIAX for Voip calls ,no hardware used yet
> I am using MeetMe to achieve conferencing  and am having a lot of delays.
> Can anyone tell me how to reduce the delay

What version of Zaptel are you using, what version of Asterisk, and
which Linux kernel does your system have?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] extensions.conf

[Moises Silva]

No limit in code imposed. Not sure about performance penalty for a
file that big, have you considered using ARA (Asterisk Realtime
Architecture)?

On 13 Jun 2006 21:06:52 +0200, andrutto <[EMAIL PROTECTED]> wrote:


Hi

Does anyone know how big extensions.conf can be?
I am trying to set up Asterisk which will have about 45 000 lines in 
extensions.conf. Is there any limitation about the amount of lines in that file?

Cheers

Andrutto


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[Asterisk-Users] Re: AGI Stderr

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Frederic Jean <[EMAIL PROTECTED]> wrote:
> 
> , never tried it with asterisk but you could redirect STDERR to STDOUT
> and see how you can capture this guy afterward...
> 
> open STDERR, ">&STDOUT";
> 
> just a thought

I think you will find that's no good, because AGI uses stdout to talk to
Asterisk.

However, to reply to Doug, if you want to log the stderr with timestamps
via syslog you could do it by piping through "logger" (man logger):

open STDERR, "| /usr/bin/logger -p local0.notice -t AGI";

Cheers
Tony
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RE: [Asterisk-Users] Hard drive write cache

[shadowym]

Thanks for the suggestions.

CF is not an option for FreePBX which is a requirement for the installs I
have in mind.  Astlinux on CF is a great option otherwise.  That is by far
the simplest, cheapest, and suprisingly most reliable solution I have come
across so far.  If there was a half decent (open source) GUI that could run
on Astlinux on CF it would be a no brainer IMHO.

Physically locking down the server is not an option.  It will be hung on the
wall in place of where a traditional PBX would normally go.  This is a
telecom closet NOT a server rack environment.  UPS with auto shut down is
just one link in the chain.  Do you have any further information of locking
plugs?  I have not come across those before.  Of course in order for that to
make sense I would need locking plugs on both the server AND UPS end.  It
has to be idiot proof.

Think PBX and/or network appliance not computer server.  They are idiot
proof so it is quite reasonable IMHO to expect the same from an Asterisk
server (or in my way of thinking, Asterisk network appliance).

> -Original Message-
> From: Nicholas Kathmann 
> [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, June 13, 2006 11:04 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Hard drive write cache
> 
> If all you are worried about is the write cache on the disks, 
> why not just put the system on a UPS set to shutdown the 
> system in the event of power failure, then place both the UPS 
> and asterisk servers in a locked rack.  In the event of a 
> power failure (or someone knocking the plug loose, which you 
> can use locking plugs to further mitigate), the system will 
> stay up on battery power then shut itself down to prevent 
> data corruption.  I doubt you will get that level of uptime, 
> but there are other options to help achieve higher 
> reliability.  You can run the OS and asterisk on a solid 
> state disk, and have voicemail and whatever else you want to 
> go to rotating disks.  That will also help with power usage 
> on the server when using the UPS.  Industrial flash disks are 
> said to have (but they really can't promise this) a 3 million 
> hour MTBF.
> 
> Thanks,
> Nick
> 
> shadowym wrote:
> > The cold hard truth is that if Asterisk cannot achieve 
> 99.999% uptime 
> > without becoming much more expensive that a traditional PBX 
> then it is 
> > not a viable alternative.  Even elcheapo Key systems are 
> rated for five nines.
> > That is what the telco world requires unless your just 
> using Asterisk 
> > in your basement as a hobby or as a one man company.
> >
> > Redundant Servers is moving into the realm of non-competitive with 
> > Traditional PBX IMHO.
> >
> > I don't care about corruption of the CDR or any of the 
> > logging/database information.  All I care about is the ability make 
> > phone calls after power failure.  That IS the MAIN function 
> of a PBX.  
> > Not call centers, databases, CDR, click 2 call, and all the 
> other bells and whistles.
> >
> >  
> >
> >   
> >> -Original Message-
> >> From: Boris Bakchiev [mailto:[EMAIL PROTECTED]
> >> Sent: Tuesday, June 13, 2006 2:13 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: RE: [Asterisk-Users] Hard drive write cache
> >>
> >> These days you don't have to worry much about your write 
> cache unless 
> >> you're running application where once single byte changed 
> will affect 
> >> whole file.
> >>
> >> Look at it this way, the only corruption will occur is 
> whatever the 
> >> files were open by asterisk at the time of the crash. And 
> only up to 
> >> the point where the file was last open.
> >> As far as I know asterisk does not keep cdr or log files 
> open so you 
> >> would loose only the data that was written at the time of 
> the power 
> >> failure.
> >>
> >> Any journaling file system (ext3, resierfs, xfs, etc) will easily 
> >> handle any power failure event. Your files will not be corrupt but 
> >> could miss some of the data.
> >>
> >> At the most you will loose 10-50 cdr entries written to 
> you log files.
> >>
> >> If you post CDR to a remote SQL database then you asterisk install 
> >> and linux is more or less static and will not be affected by the 
> >> power failure.
> >>
> >> What you need to do is minimise the writes to hard disk's:
> >>
> >> 1 - Send syslog to remote server and do not do ANY syslogs
> >> Or keep the circular buffer in memory if you have 
> plenty of it. 
> >> 2 - Send CDR's to SQL server (or log to ramdisk and send to remote 
> >> server every few minutes via SSH)
> >> 3 - Do not record any calls (or do that somewhere else)
> >> 4 - Stop any services that write/read data on regular intervals.
> >>
> >> If you have no writes you have nothing to worry about during power 
> >> failure and journaling file system will take care of the rest.
> >>
> >> Keep your partition size really small so that fsck will 
> not take much 
> >> time.
> >>
> >> You ha

Re: [Asterisk-Users] Can this config sustain 30 users?

[Mike Fedyk]

Martin Joseph wrote:
Ultimately you need to set up a server that does what you need and see 
how it performs. Usually hardware overkill is a good bet,  but you 
don't need to go crazy.

So, one cpu per call is too much?
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Re: [Asterisk-Users] Can this config sustain 30 users?

[TSS Support]

Just so you know ... you can't just add up data transfer like that. Also, 
you're dealing with best-case link speed.


Anyway, all of this is sort of meaningless. At high non-transcoded call 
volumes the following sorts of questions are more relevent:


-Is my NIC well supported by Linux?
-Is my disk chipset/controller well supported by Linux?
-Are parts common enough that I can get them quickly when one fails, or will 
I keep spares on hand?


-Keith

- Original Message - 
From: "Erick Perez" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 13, 2006 8:14 AM
Subject: Re: [Asterisk-Users] Can this config sustain 30 users?

Means that the cpu,the ram and the board can achieve (see point b)
about 34 gigabits of data transfer, but 24 users only generate 3.75
megabits. So this is more than covered.
However if we take into account the lowest performing component on
this system (the sata disks) we go down to 1.5gbits/s which still
seems to be enough.


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Re: [Asterisk-Users] Can this config sustain 30 users?

[Martin Joseph]

On Jun 13, 2006, at 10:15 AM, Erick Perez wrote:

So if tomorrow i need 50, 60 or 100 users, I just need to reinvent the 
wheel

every time.



It's not that simple.  Every system has different requirement and 
configuration, and therefore load is different.  Some people record 
lots of calls or use G729, this changes the maximum # of users 
radically.


Transcoding of music or voice mail also adds load.

Bandwidth constraints are a major factor for SIP or IAX call 
termination.


Ultimately you need to set up a server that does what you need and see 
how it performs. Usually hardware overkill is a good bet,  but you 
don't need to go crazy.


Marty


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[Asterisk-Users] extensions.conf

[andrutto]

Hi

Does anyone know how big extensions.conf can be?
I am trying to set up Asterisk which will have about 45 000 lines in 
extensions.conf. Is there any limitation about the amount of lines in that file?

Cheers

Andrutto


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[Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing

[BerkHolz, Steven]

 
 I edited the calleridname.agi patch to only overwrite the name if it is
missing.
The asteridex option still overwrites the name since it is our master
list for known numbers.

-- 

Steven


calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue
Jun 13 14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13
14:37:09 2006
@@ -16,6 +16,7 @@

 my $callerid = $input{'calleridnum'};
 my $calleridfull = $input{'callerid'};
+my $calleridname = $input{'calleridname'};

 if($callerid eq ''){
 $callerid=$input{'callerid'};
@@ -31,7 +32,8 @@

 $calleridfull =~ s/[\,\"\']+/ /g;

-$AGI->verbose("CALLERID IS: $calleridfull\n");
+$AGI->verbose("CALLERID IS: $calleridfull\n");
+$AGI->verbose("CALLERID Name IS: $calleridname\n");

 if ($callerid =~ /^(\d{3})(\d{3})(\d{4})$/) {
 $npa = $1;
@@ -54,7 +56,7 @@
 #$nxx='892';
 #$station='8019';

-if ($Fonetastic > '0') {
+if (($Fonetastic > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for Fonetastic.US lookup... \n");
 if ($name = &fonetastic_lookup ($npa, $nxx)) {
  $newcallerid = "\"$name <$npa$nxx$station>\"";
@@ -68,7 +70,7 @@
 $AGI->verbose("Fonetastic.US lookup disabled.");
 }

-if ($AnyWho > '0') {
+if (($AnyWho > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for AnyWho lookup... \n");
 if ($name = &anywho_lookup ($npa, $nxx, $station)) {
 $newcallerid = "\"$name <$npa$nxx$station>\"";
@@ -82,7 +84,7 @@
 $AGI->verbose("AnyWho lookup disabled.");
 }

-if ($Google > '0') {
+if (($Google > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for Google lookup... \n");
 if ($name = &google_lookup ($npa, $nxx, $station)) {
 $newcallerid = "\"$name <$npa$nxx$station>\""; 


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Re: [Asterisk-Users] echo sidetone grandstream and tdm400p

[Martin Joseph]

On Jun 13, 2006, at 9:54 AM, Marco Sajeva wrote:


First, thank you for your quick and kind answer.
I cannot change the TX gain on the Grandstream phones, or atleast I 
don't know

how to...
Can anybody help, please?
Actually since you suggested the problem is only with your PSTN calls 
(ie Zap channels), I think the answer is more likely in the 
zapata.conf?


I would experiment a bit more there with the gains...

Marty


Thanks in advance,
Marco


On Tue, 13 Jun 2006 08:54:31 -0600, Colin Anderson wrote
Turn down your microphone TX gains on the phones. On my TDM400 with 
Vista

350's I had to crank the mic value way down. This is not specific to
FXS phones, on my Snom 200's sidetone is so bad, that an appropriate
setting for mic gain is '2' (out of 8)

hth

-Original Message-
From: Marco Sajeva [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 8:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] echo sidetone grandstream and tdm400p

Hi all,
thanks to the all of you. This list is very interesting also for a 
newby

like me.
My problem: I just setup my first full working asterisk installation
with this config:
1. n.1 GXP-2000
2. n.4 Budgetone 102
3. n.1 TDM400p (3 FXS, 1 FXO)

Everything seems to work fine, but the sidetone... it's really 
annoying!

We can hear the sidetone only when we call to the outside (PSTN), it
doesn't matter if we call a local, a mobile or a longdistance call.
Only we hear the echo, not the called party. We do not ear any echo
in internal call to each other extensions. I tryed every possible
setting of the echotraining, of the rx and of the tx gain, but with
no success. Any idea or help? Thank you in advance, Marco

__
Dott. Ing. Marco Sajeva
Visioni - we network
http://www.visioni.info
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Re: [Asterisk-Users] No incoming sip calls

[Russell Horn]

On 6/13/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote:

Could your register line require attention ? (2001?)

 7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201?


That's a good spot and I've fixed it now, but I'm sure it's not the
problem. I'm not seeing any sip traffic coming in at all, I'd have
expected if I jsut had the wrong extensions to have seen both traffic
and errors at the console.

Thanks though - I'll keep looking.

Russell
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Re: [Asterisk-Users] Voiicemail NFS Cutting Out

[Martin Joseph]

On Jun 13, 2006, at 10:08 AM, BILL GITONGA wrote:



Hi,

I have two asterisk systems that share voicemail on an
NFS. I recently upgraded to Asterisk 1.2.9.1.
After the upgrade, the voicemail gets cut out after
about 5 seconds of recording. Any ideas on what might
be causing this?

I think there was a "fix" for silence detection, that is just a wild 
guess on my part of course...


Marty

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[Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing

[Steven]

I edited the calleridname.agi patch to only overwrite the name if it is missing.
The asteridex option still overwrites the name since it is our master list for 
known numbers.

-- 

Steven
calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 
14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 
14:37:09 2006
@@ -16,6 +16,7 @@

 my $callerid = $input{'calleridnum'};
 my $calleridfull = $input{'callerid'};
+my $calleridname = $input{'calleridname'};

 if($callerid eq ''){
 $callerid=$input{'callerid'};
@@ -31,7 +32,8 @@

 $calleridfull =~ s/[\,\"\']+/ /g;

-$AGI->verbose("CALLERID IS: $calleridfull\n");
+$AGI->verbose("CALLERID IS: $calleridfull\n");
+$AGI->verbose("CALLERID Name IS: $calleridname\n");

 if ($callerid =~ /^(\d{3})(\d{3})(\d{4})$/) {
 $npa = $1;
@@ -54,7 +56,7 @@
 #$nxx='892';
 #$station='8019';

-if ($Fonetastic > '0') {
+if (($Fonetastic > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for Fonetastic.US lookup... \n");
 if ($name = &fonetastic_lookup ($npa, $nxx)) {
  $newcallerid = "\"$name <$npa$nxx$station>\"";
@@ -68,7 +70,7 @@
 $AGI->verbose("Fonetastic.US lookup disabled.");
 }

-if ($AnyWho > '0') {
+if (($AnyWho > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for AnyWho lookup... \n");
 if ($name = &anywho_lookup ($npa, $nxx, $station)) {
 $newcallerid = "\"$name <$npa$nxx$station>\"";
@@ -82,7 +84,7 @@
 $AGI->verbose("AnyWho lookup disabled.");
 }

-if ($Google > '0') {
+if (($Google > '0') && ($calleridname != 'unknown')){
 $AGI->verbose("Ready for Google lookup... \n");
 if ($name = &google_lookup ($npa, $nxx, $station)) {
 $newcallerid = "\"$name <$npa$nxx$station>\""; 



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Re: [Asterisk-Users] No incoming sip calls

[Jonathan Attwood]

Could your register line require attention ? (2001?)

7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201?

On 13/06/06, Russell Horn <[EMAIL PROTECTED]> wrote:

Hi folks - I've recently returned to asterisk after an eighteen month break.

I've two sip providers - gradwell in the UK (inbound and outbound)
and talklite in the US (outbound only).

I've managed to get outbound dialing working but am not receiving any
calls from gradwell.

I've included my sip.conf and extensions.conf as well as the output
from tethereal. When a call is placed to rgadwell I'm seeing no sip
traffic whatsoever on asterisk. My aim is to have inbound calls ring
SIP extension 2201

I'm guessing this is something pretty straightforward, but any help
would be much appreciated.

Thanks,

Russell.

sip.conf

[general]
context=incoming; Default context for incoming calls
register => 7960xxx:[EMAIL PROTECTED]/2001
register => 9479xxx:[EMAIL PROTECTED]
port=5060   ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all)
nat=yes ; NAT settings
allow=all

[Gradwell]
type=peer
username=796
fromuser=796
secret=
host=sip.gradwell.net
context=flat
fromdomain=sip.gradwell.net
nat=yes
allow=all
canreinvite=no
dtmfmode=inband
qualify=yes

[talklite]
type=peer
username=9479
qualify=yes
secret=
host=sip.talklite.net
canreinvite=yes
disallow=all
allow=ulaw

[2201]
type=friend
context=flat
username=albanach
secret=
defaultip=192.168.1.100
qualify=yes
type=friend
callerid="Russell Horn" <>
host=dynamic
nat=no   ; X-Lite is behind a NAT router
canreinvite=yes   ; Typically set to NO if behind NAT
allow=all


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

extensions.conf

[general]
static=yes
writeprotect=no

[globals]
TRUNK=Gradwell
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

PHONES1=SIP/2201


[flat]
include => home
include => outgoing

[home]
exten => 2201,1,Dial(${PHONES1},20,Ttm)
exten => 2201,2,Macro(vmessage,${PHONES1VM})
exten => 2201,3,Hangup

[outgoing]
ignorepat => 9
ignorepat => 8
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

linux:/etc/asterisk # tethereal -R "sip"
Capturing on eth0
 0.00 207.44.248.78 -> 192.168.1.102 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]
 0.000831 192.168.1.102 -> 207.44.248.78 SIP Status: 404 Not Found
 1.350584 192.168.1.102 -> 192.168.1.100 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]:5060
 1.350730 192.168.1.102 -> 207.44.248.78 SIP Request: OPTIONS
sip:sip.talklite.net
 1.350887 192.168.1.102 -> 193.111.200.56 SIP Request: OPTIONS
sip:sip.gradwell.net
 1.369388 192.168.1.100 -> 192.168.1.102 SIP Status: 200 OK
 1.455492 207.44.248.78 -> 192.168.1.102 SIP Status: 404 Not Found
 1.502618 193.111.200.56 -> 192.168.1.102 SIP Status: 404 Invalid
account for voicemail
 1.552845 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
 1.654933 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying(1 bindings)
 1.655832 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
 1.657951 207.44.248.78 -> 192.168.1.102 SIP Status: 401 Unauthorized
  (1 bindings)
 1.658229 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
 1.770875 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying(1 bindings)
 1.773894 207.44.248.78 -> 192.168.1.102 SIP Status: 200 OK(1 bindings)
 1.792718 193.111.200.56 -> 192.168.1.102 SIP Status: 401
Unauthorized(0 bindings)
 1.793529 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
 1.937253 193.111.200.56 -> 192.168.1.102 SIP Status: 200 OK(1 bindings)
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Re: [Asterisk-Users] Can this config sustain 30 users?

[Noah Miller]

Hi Erick -


So at the end, i cannot provide prices (without being overkill) to a
potential customer without spending money on a system prior to quote
and test it.


You can always do what you just did and ask on the list.

I think the real reason there is no "table" or chart for hardware
specs is that there are just too many things you can do with asterisk.
What connection technologies are you using?  IAX, SIP, MGCP, H323,
Skinny?  Are you transcoding?  What kind of transcoding?  Are you
using PSTN Hardware Cards?  How many channels?  Are you recording
calls?  Are you using meetme?  All these questions (and MANY more) are
relevant, and for each different answer you can potentially have
radically different hardware specs.

I've found that many software vendors neglect to provide hardware
specs for their server software.  Digium/Asterisk is certainly not
alone in this.  Even hardware vendors are guilty - try asking Cisco
how much RAM or CPU you need in a router in order to do a particular
task.

You're note completely in the dark either.  Many folks have done
scalability testing with asterisk and have published their results.
It's easy enough to read those results and make comparisons to what
you'll be doing with asterisk.  I also think it's important to note
that asterisk has been used in enterprises since at least 2002, and no
hardware from then can hold a candle to 2006 hardware.

- Noah
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Re: [Asterisk-Users] Snom high SIP ping time

[Steven Ringwald]

Steve Glaus wrote:

Mike Hammett wrote:
I don't know everything that's going on as someone else has been 
working on the project, but it hasn't really been going anywhere, so 
I had some questions.
 
We've got some Snom 320s with Asterisk 1.2.9.1 (I believe).  All was 
well (with a previous release), but the phones started to get real 
choppy.  We are also running a softphone at this location and it was 
fine.  The SIP qualify was returning ping times anywhere from 20 to 
70 ms over a sparsely used LAN.  Command prompt (ICMP) pings were 
under 1 ms.  No amount of different Asterisk versions or phone 
firmware revisions seems to solve this.  All was well, then (as far 
as we know) without changes, it crapped out.
  
I'm having much the same issues only I'm using Cisco 7960 phones. When 
I do a 'sip show peers' I'm getting times in excess of 300ms. A soft 
phone on the same network (x-lite), is reporting times of 4 ms. 
Related to this (I think), I'm getting audio issues. The person being 
called can hear the caller fine but the callee's voice drops in and 
out excessively.


I have qualify set to yes in the sip definitions for all the clients 
(Including the soft phone). Does anyone know what is causing this. I'm 
not aware what the sip ping times were earlier, but the audio issues 
seemed to have started spontaneously.


Anyone  have any idea regarding this? 



What codecs are you using? I have noticed that g729, for some reason, 
adds a lot of latency to the phone. Running on uLaw, however, I get 
times from sip show peers of around 5-14ms.


Steve

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Re: [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?

[Jonathan Attwood]

have a look here 
for a decent write up, from a Noth American perspective, about wirng
VoIP into your home.

Beware, the pages at voxilla can take a while to load

On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote:

Ahh, thanks!  That's what I thought but I wasn't sure because I
thought ATA boxes were only for specific VOIP providers.

Which ATA with an FXS port would you recommend for around (or under) $50?

Also, can I simple plug the ATA into an existing RJ-11 jack so that
ALL of the phone jacks in my house have a dial tone?


On 6/12/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote:
> Analogue Telephone Adapter(s)
> Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3
>
> On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
> > Ok, I've done some more research and I don't think I want an FXO box...
> >
> > What I'd like to do is use BroadVoice (with their BYOD plan) and then
> > run Asterisk on my WRT54G router. I'd also like to use my regular home
> > phones without having to use a special "SIP" phone... (eg. I like my
> > Vtech normal cordless phones)
> >
> > What do I need to buy to get this working? It sounds like I need to
> > purchase a "Zaptel" interface card, but of course I can't use those
> > with a router...
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RE: [Asterisk-Users] Asterisk & Eyebeam chat function

[Douglas Garstang]

No problem.
SER and OpenSER do support MESSAGE though... 

> -Original Message-
> From: Attilla De Groot [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, June 13, 2006 11:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk & Eyebeam chat function
> 
> 
> Hi Doug,
> 
> 
> I didn't knew this.
> Thank you.
> 
> 
> Regards,
> Attilla
> 
> On Jun 13, 2006, at 4:52 PM, Douglas Garstang wrote:
> 
> > Unless it's changed recently, Asterik doesn't support the SIP  
> > 'MESSAGE' command.
> >
> > Doug.
> >
> >> -Original Message-
> >> From: Attilla De Groot [mailto:[EMAIL PROTECTED]
> >> Sent: Tuesday, June 13, 2006 2:41 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: [Asterisk-Users] Asterisk & Eyebeam chat function
> >>
> >>
> >> Hi all,
> >>
> >>
> >> Eyebeam has a sip-chat function and it would be nice if I would be
> >> able to use it. But the problem is that I can't really find
> >> information about it.
> >>
> >> I can just try to send a message and on the Asterisk console a
> >> message like this appears:
> >>
> >> Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message:
> >> Received message to  from "Bla
> >> Sheep";tag=1d072048, dropped it...
> >>Content-Type:text/plain
> >>Message: ?
> >>
> >> Can anyone tell me more about this or give me a link with some
> >> information about it ?
> >>
> >>
> >> Regards,
> >> Attilla de GrootÎ
> >>
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[Asterisk-Users] Grandstream BT101 Auto-Answer

[Jon Scottorn]

Hi,

   I am wondering if anyone has gotten the BT101's to work with the paging in Asterisk?  I know that the phones themselves have an auto-answer option and if I turn it on every call is auto answered.  I want to be able to call the extension normally and have it ring normally but if someone dials # and the extension to have it auto answer for intercom purposes.

Anyone have this working?

Thanks in advance for any advise or remarks.




Jon Scottorn





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Re: [Asterisk-Users] Hard drive write cache

[Nicholas Kathmann]

If all you are worried about is the write cache on the disks, why not 
just put the system on a UPS set to shutdown the system in the event of 
power failure, then place both the UPS and asterisk servers in a locked 
rack.  In the event of a power failure (or someone knocking the plug 
loose, which you can use locking plugs to further mitigate), the system 
will stay up on battery power then shut itself down to prevent data 
corruption.  I doubt you will get that level of uptime, but there are 
other options to help achieve higher reliability.  You can run the OS 
and asterisk on a solid state disk, and have voicemail and whatever else 
you want to go to rotating disks.  That will also help with power usage 
on the server when using the UPS.  Industrial flash disks are said to 
have (but they really can't promise this) a 3 million hour MTBF.


Thanks,
Nick

shadowym wrote:

The cold hard truth is that if Asterisk cannot achieve 99.999% uptime
without becoming much more expensive that a traditional PBX then it is not a
viable alternative.  Even elcheapo Key systems are rated for five nines.
That is what the telco world requires unless your just using Asterisk in
your basement as a hobby or as a one man company.

Redundant Servers is moving into the realm of non-competitive with
Traditional PBX IMHO.

I don't care about corruption of the CDR or any of the logging/database
information.  All I care about is the ability make phone calls after power
failure.  That IS the MAIN function of a PBX.  Not call centers, databases,
CDR, click 2 call, and all the other bells and whistles.

 

  

-Original Message-
From: Boris Bakchiev [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 13, 2006 2:13 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Hard drive write cache

These days you don't have to worry much about your write 
cache unless you're running application where once single 
byte changed will affect whole file.


Look at it this way, the only corruption will occur is 
whatever the files were open by asterisk at the time of the 
crash. And only up to the point where the file was last open. 
As far as I know asterisk does not keep cdr or log files open 
so you would loose only the data that was written at the time 
of the power failure.


Any journaling file system (ext3, resierfs, xfs, etc) will 
easily handle any power failure event. Your files will not be 
corrupt but could miss some of the data.


At the most you will loose 10-50 cdr entries written to you log files.

If you post CDR to a remote SQL database then you asterisk 
install and linux is more or less static and will not be 
affected by the power failure.


What you need to do is minimise the writes to hard disk's:

1 - Send syslog to remote server and do not do ANY syslogs
Or keep the circular buffer in memory if you have plenty of it. 
2 - Send CDR's to SQL server (or log to ramdisk and send to 
remote server every few minutes via SSH)

3 - Do not record any calls (or do that somewhere else)
4 - Stop any services that write/read data on regular intervals.

If you have no writes you have nothing to worry about during 
power failure and journaling file system will take care of the rest.


Keep your partition size really small so that fsck will not 
take much time.


You have to be realistic, you cannot achieve 99.999% uptime. 
That's 5 minutes per year downtime.

You will have more or less 100% until your first hardware failure.

Even if you have all the hardware components pre-purchased it 
will still take you 2-12 hours to detect, diagnose and fix 
the fault if you lucky.
So your 5 minuets 

If the business is demanding 99.999% then it should be 
prepared to invest into the hardware.

I would recommend a cluster or even better a fault tolerant server.
Those are expensive but you can pretty much rule out the 
hardware failure and swap all of the failed components while 
the system is running (cpu, memory, hdd, etc).


Look at Stratus or NEC FT servers if you need hardware redundancy.
They're expensive but will give you the hardware reliability you need.

Or get a traditional PABX :)





-Original Message-
From: [EMAIL PROTECTED] 
  
[mailto:asterisk-users- 


[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Tuesday, 13 June 2006 10:34
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hard drive write cache


I am looking at ways to harden my asterisk install to 
  
prevent computer 

related issues from happening.  I am concerned about about 
  
disk write 


cache.
That seems to be a major source of hard drive corruption on power
  

failure.

Hard Drive corruption is simply unacceptable for the 99.999% uptime 
requirements of my Asterisk install that needs to be as 
  
reliable as a 


proprietary PBX.

Of course I will be using redundant power supplies, raid 1 and use a
  

UPS.


None of those things mean much if the power 

Re: [Asterisk-Users] transferring calls from ekiga to asterisk

[don Paolo Benvenuto]

El mar, 13-06-2006 a las 07:33 +, [EMAIL PROTECTED]
escribió:
> When you configured the incoming line in sip.conf, you gave it a context.

I think my problem is: how do I configure sip.conf in order to receive
those call redirects?

In Twinklephone I have two accounts:
- an account in which twinkle is a peer of asterisk, which has this
settings in sip.conf:

[pablopctwinkle]
type=friend
secret=xx
callerid="Pablo PC Twinklephone" <619>
host=dynamic
context=todo
nat=no
qualify=yes

twinkle registers with asterisk without problems with these settings. It
sends and receives calls, it's a normal asterisk's extension

- another with the voip provider (voip.eutelia.it)

This account is the one that receives the calls and redirects it to
asterisk.

I want to transfer a call from this account to asterisk. That's
equivalent, I think, to connecting to asterisk from that account.

196.3.84.214 my routers external address
5062 is the port that twinklephone uses
10.152.58.1=misiongenovesa is the server asterisk and twinklephone are
running on
0108937227 is my username with voip provider voip.eutelia.it

I must configure sip.conf and extensions.conf in order to receive calls
from that account.

If I try to call asterisk from that account I get in asterisk's console
(sip debug):

---BEGIN-
<-- SIP read from 10.152.58.1:5062:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: 
From: "don Paolo Benvenuto" ;tag=cfpll
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (13 headers 14 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 196.3.84.214 : 5062 (NAT)
Found peer 'pablopctwinkle'
Reliably Transmitting (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="78099a8f"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000
ms
Retransmitting #1 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="78099a8f"
Content-Length: 0


---
Retransmitting #2 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="78099a8f"
Content-Length: 0


---
Retransmitting #3 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="78099a8f"
Content-Length: 0


---
Retransmitting #4 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="78099a8f"
Content-Length: 0


---
Retransmitting #5 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto" ;tag=cfpll
To: ;tag=as111cfbda
Call-ID: [EMAIL PROTECTED]
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Proxy-Authenticate: Digest

Re: [Asterisk-Users] What is Echo?

[C F]

Echo is when you throw a basket ball on the floor and it bounces back,
the effect of the ball coming back to you is called Echo. If you go
into an empty big room and yell out I hate clinton you should hear the
walls agreeing with you and thats called echo.

On 6/13/06, Crazy Boy <[EMAIL PROTECTED]> wrote:

Hi Friend,

I heard about this word "echo" very much. Can you please tell me what is
this "Echo"?

Thanks&Regards,
Chandra


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Re: [Asterisk-Users] Snom high SIP ping time

[Steve Glaus]

Mike Hammett wrote:
I don't know everything that's going on as someone else has been 
working on the project, but it hasn't really been going anywhere, so I 
had some questions.
 
We've got some Snom 320s with Asterisk 1.2.9.1 (I believe).  All was 
well (with a previous release), but the phones started to get real 
choppy.  We are also running a softphone at this location and it was 
fine.  The SIP qualify was returning ping times anywhere from 20 to 70 
ms over a sparsely used LAN.  Command prompt (ICMP) pings were under 1 
ms.  No amount of different Asterisk versions or phone firmware 
revisions seems to solve this.  All was well, then (as far as we know) 
without changes, it crapped out.
 
Any ideas?
 
 


Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
 
 



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I'm having much the same issues only I'm using Cisco 7960 phones. When I 
do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone 
on the same network (x-lite), is reporting times of 4 ms. Related to 
this (I think), I'm getting audio issues. The person being called can 
hear the caller fine but the callee's voice drops in and out excessively.


I have qualify set to yes in the sip definitions for all the clients 
(Including the soft phone). Does anyone know what is causing this. I'm 
not aware what the sip ping times were earlier, but the audio issues 
seemed to have started spontaneously.


Anyone  have any idea regarding this?
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Re: [Asterisk-Users] GXP-2000

[Daniel Salama]

Would you mind telling me how to setup the GXP-2000's VLAN/QoS  
settings with the DES-1226G? I just purchased the DES-1226G and want  
to make sure I setup it up right. I don't have the ability to run  
separate wiring for the PC and the phone and that's why I need this  
help.


Thanks,
Daniel

On Jun 7, 2006, at 9:52 PM, Mike Fedyk wrote:

I have heard good things about the D-Link DES-1226G switch ($150 at  
newegg).  If you can run a separate cable to the computer and  
phone.  If you can't run the extra cables, then configure your  
phone to tag itself as part of the voip vlan and let the switch tag  
everything else as the computer vlan.


I happen to have asterisk running as a router, so I use it doing  
QoS with tc (traffic control) and wondershaper set to prioritize  
based on port ranges.  I sent a patch to the debian bug tracking  
system a while back with a few improvements -- I should check on  
that.  It basically prioritizes smaller packets before larger  
packets with ~8 levels of priority and groups of sizes for the  
packets.  Just doing that automatically handles 80% of the need for  
prioritization without specifying port ranges for the sip rtp packets.


Mike

Daniel Salama wrote:
They are extremely casual web surfers. Just have their Outlook  
client opened checking email every minute. Email traffic is very low.


They are all connected to the same switch. It's a Netopia DSL  
router/modem/switch for the BellSouth DSL service. The computers  
are connected to the PC port behind the GXP-2000.


Any suggestions?

Thanks,
Daniel

On Jun 7, 2006, at 8:49 PM, list mail wrote:

What do they do on the internet? Heavy surfing, large transfers,  
myspace. How are these units connected to the network? Are they  
passing through the same switch?

I don't think it is the phones...

On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:


Mike,

I added a qualify=500 on those phones. My client has peers  
100218 thru 100222 (a total of 5 phones). Below is the messages  
log since I activated it this morning at 8:30AM:


Jun  7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1075ms / 500ms)
Jun  7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (66ms / 500ms)
Jun  7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1075ms / 500ms)
Jun  7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (68ms / 500ms)
Jun  7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now  
TOO LAGGED! (1114ms / 500ms)
Jun  7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now  
REACHABLE! (90ms / 500ms)
Jun  7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now  
TOO LAGGED! (1077ms / 500ms)
Jun  7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now  
REACHABLE! (72ms / 500ms)
Jun  7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now  
TOO LAGGED! (1077ms / 500ms)
Jun  7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now  
REACHABLE! (73ms / 500ms)


As you can see, it only happens to a couple of their phones and  
at random times. They're behind a DSL circuit. I don't know if  
it's because their DSL line is going up/down. They don't  
necessarily claim their Internet goes down, however, they are  
not constantly check it.


What would you (or anyone else) suggest?

Thanks,
Daniel

On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:

Do you have multiple phones going down at the same time?  If  
so, monitor them with "qualify=500" in sip.conf to see if they  
hit that limit.  If you see more than one go down within a  
short period of time, you have network problems.  Check the  
quality of the network switches they have.


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[Asterisk-Users] Voice Mail NFS Problem-Cutting Out

[BILL GITONGA]

I have two asterisk systems that share voicemail on an
NFS. I recently upgraded to Asterisk 1.2.9.1. 
After the upgrade, the voicemail gets cut out after
about 5 seconds of recording. Any ideas on what might
be causing this?


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