[asterisk-users] asterisk-users Digest, Vol 37, Issue 88

Wed Aug 22 20:57:11 CDT 2007

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: Wednesday, August 22, 2007 10:51 PM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 37, Issue 88

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Today's Topics:

   1. Re: 99 bottles of beer (David Cook)
   2. DUNDi, So Easy A Caveman Could Do It! (JR Richardson)
   3. Polycom behind NAT won't register to * server	behind ALG
      (Matthew Warren)
   4. Re: Polycom behind NAT won't register to * server behind ALG
      (Alex Balashov)
   5. Re: Polycom and NAT (Darryl Dunkin)
   6. Re: Polycom behind NAT won't register to * server behind ALG
      (Henry L.Coleman)
   7. Re: Polycom behind NAT won't register to *	serverbehind ALG
      (Marty Mastera)
   8. rfc3680, reginfo+xml (Olivier)
   9. How to re-read values from database in Trixbox (Edgar Guadamuz)
  10. Re: How to re-read values from database in Trixbox
      (Diego Iastrubni)
  11. Re: Saftware RAID1 or Hardware RAID1 with Asterisk
      (Richard Scobie)
  12. How do I configure asterisk? (fateme fatah)
  13. Which interface? (fateme fatah)
  14. Re: rfc3680, reginfo+xml (Raj Jain)
  15. Cisco firmwares 3.6.3 vs 3.8.6 (Adrian Marsh)
  16. Re: compatibility of PRI Two B channel transfers	TBTC/2BTC
      (Matt Florell)
  17. Re: DUNDi, So Easy A Caveman Could Do It! (Lenz)
  18. Re: Cisco firmwares 3.6.3 vs 3.8.6 (Arnaud Ligot)
  19. Re: rfc3680, reginfo+xml (Olivier)
  20. asterisk with FAX problem (satish patel)
  21. Re: Polycom and NAT (Klaverstyn, David C)
  22. Re: How do I configure asterisk? (Atis)
  23. Re: Polycom behind NAT won't register to * server behind ALG
      (Eric "ManxPower" Wieling)
  24. Re: 99 bottles of beer (Russell Handorf)
  25. Re: Saftware RAID1 or Hardware RAID1 with Asterisk (Steven)

----------------------------------------------------------------------

Message: 1
Date: Tue, 21 Aug 2007 21:01:50 -0400
From: "David Cook" <dbc_asterisk at advan.ca>
Subject: Re: [asterisk-users] 99 bottles of beer
To: <asterisk-users at lists.digium.com>
Message-ID: <012501c7e458$05919370$10b4ba50$@ca>
Content-Type: text/plain; charset="us-ascii"

On 8/21/07, Steve Edwards <asterisk.org at sedwards.com> wrote:

> 

> "To control the tv in this room, press 1. To control a tv in another

> room, press 2. To control the outside lights, press 3. To control the

> sprinklers, press 4, ..."

> 

Before this thread I already had a Firecracker on the server, a fair
assortment of lights and the sprinklers are on an X10Pro Irrigation
Controller.

Damn, now I'm gonna be up all night.....

- dbc.

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Message: 2
Date: Tue, 21 Aug 2007 20:51:51 -0500
From: "JR Richardson" <jmr.richardson at gmail.com>
Subject: [asterisk-users] DUNDi, So Easy A Caveman Could Do It!
To: asterisk-users at lists.digium.com, dundi at lists.digium.com
Message-ID:
	<4a28136b0708211851u44938163u4ae6a83051c8b4af at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Here you go folks:

ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf

If someone would be so kind as to upload to the wiki, it will be much
appriciated.

Thank you all who replied to my poll questions.

As always, I hope this help.

JR
--
JR Richardson
Engineering for the Masses

------------------------------

Message: 3
Date: Tue, 21 Aug 2007 22:03:30 -0400
From: "Matthew Warren" <mwarren at ru-intouch.com>
Subject: [asterisk-users] Polycom behind NAT won't register to *
	server	behind ALG
To: <asterisk-users at lists.digium.com>
Message-ID: <JDEJIIDIEODAPFNPEOAAOELFEFAA.mwarren at ru-intouch.com>
Content-Type: text/plain;	charset="us-ascii"

Polycom's were simply not originally built for multi location VoIP.  There
is no NAT support in the Polycom's. We have several networks, being an ISP,
and have found that when transversing one network say 192.168.2.x with the *
box on a 192.168.1.x the polycoms were able to communicate however sustained
a lot of one way audio problems.  Moving thim onto the same network is the
only thing we have been able to reliable do.  According to Polycom Support
this is what they are intended for and no definitive answer as to whether
they would support Stun or another method in the future.  At least as of 6
months ago.

Matt

------------------------------

Message: 4
Date: Tue, 21 Aug 2007 22:17:17 -0400 (EDT)
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [asterisk-users] Polycom behind NAT won't register to *
	server behind ALG
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<Pine.LNX.4.64.0708212216050.4376 at arpeggio.evaristesys.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

On Tue, 21 Aug 2007, Matthew Warren wrote:

> We have several networks, being an ISP, and have found that when 
> transversing one network say 192.168.2.x with the * box on a 192.168.1.x 
> the polycoms were able to communicate however sustained a lot of one way 
> audio problems.  Moving thim onto the same network is the only thing we 
> have been able to reliable do.

   Forgive what may be a naively misplaced line of questioning, but what 
precisely does this have to do with NAT as such?  Unless you mean to
imply otherwise, it would seem to me you are referring to 192.168.1.0/24
and 192.168.2.0/24 being intermediated by way of a router -- but not 
necessarily NAT'd?

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671

------------------------------

Message: 5
Date: Tue, 21 Aug 2007 20:24:14 -0700
From: "Darryl Dunkin" <ddunkin at netos.net>
Subject: Re: [asterisk-users] Polycom and NAT
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<EC5DC6B7B2274B47A28A30C5FCF8744001AF2C6D at exchange.nosi.netos.com>
Content-Type: text/plain; charset="us-ascii"

In your sip.conf, for the user:
nat=yes

To send keepalives for the UDP connection (depending on how flimsy the
device handling NAT is):
qualify=yes

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Klaverstyn, David C
Sent: Tuesday, August 21, 2007 17:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom and NAT

Hi All,

I have a Polycom 501 that is behind a NAT.  When it registers to the
Asterisk server it is using the IP address on the private network and
not the public IP of the NAT address.

Can someone tell me what I need to do so the phone registerers using an
internet address rather than the remote network NAT address.

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Message: 6
Date: Tue, 21 Aug 2007 23:25:22 -0400 (EDT)
From: "Henry L.Coleman" <asterisk at voip-pbx.ca>
Subject: Re: [asterisk-users] Polycom behind NAT won't register to *
	server behind ALG
To: asterisk-users at lists.digium.com
Message-ID:
	<63952.72.53.148.180.1187753122.squirrel at webmail.perimeter911.com>
Content-Type: text/plain;charset=iso-8859-1

I think what Alex was trying to say was that Polycom IP Phones are an
example of immature product development. While they look very nice and
have a nice display the product doesn't compete very well compared to
other manufacturers.
The two most obvious flaws are that they cannot be NAT'ed so they cannot
be used as Off Premise eXtensions phones and the other being that they
take so long to configure and re-boot. I have a golden rule with any phone
that I plan on installing for a customer....If I can't get it working
within 20 minutes then don't use it. I'm afraid Polycom breaks my golden
rule.
With such a lot of competition in this market they should have sorted this
out two years ago.

-- 
Henry L. Coleman.

< Alex Balashov>
> On Tue, 21 Aug 2007, Matthew Warren wrote:
>
>> We have several networks, being an ISP, and have found that when
>> transversing one network say 192.168.2.x with the * box on a 192.168.1.x
>> the polycoms were able to communicate however sustained a lot of one way
>> audio problems.  Moving thim onto the same network is the only thing we
>> have been able to reliable do.
>
>    Forgive what may be a naively misplaced line of questioning, but what
> precisely does this have to do with NAT as such?  Unless you mean to
> imply otherwise, it would seem to me you are referring to 192.168.1.0/24
> and 192.168.2.0/24 being intermediated by way of a router -- but not
> necessarily NAT'd?
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    :
> Direct :
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

------------------------------

Message: 7
Date: Tue, 21 Aug 2007 22:20:20 -0600
From: "Marty Mastera" <marty at m3resources.com>
Subject: Re: [asterisk-users] Polycom behind NAT won't register to *
	serverbehind ALG
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <E25AAE8239598347B01E9AB08EEAD0F63B858D at trajan.mastera.us>
Content-Type: text/plain;	charset="windows-1250"

> Polycom's were simply not originally built for multi location VoIP.
> There
> is no NAT support in the Polycom's. We have several networks, being an
> ISP,
> and have found that when transversing one network say 192.168.2.x with
> the *
> box on a 192.168.1.x the polycoms were able to communicate however
> sustained
> a lot of one way audio problems.  Moving thim onto the same network is
> the
> only thing we have been able to reliable do.  According to Polycom
> Support
> this is what they are intended for and no definitive answer as to
> whether
> they would support Stun or another method in the future.  At least as
> of 6
> months ago.
> 
> Matt
> 

Although I do appreciate your response, I didn't intend to paint this as a
NAT issue in my original post.  I have successfully deployed Polycom phones
behind NAT many times in the past when the * box was on a public IP without
a NAT or ALG present.  This leads me to focus on the ALG as part of issue in
this case (not that the ALG in and of itself is the issue, but the
combination of Polycom and the ALG since other brands of phones work
properly).  The link that I referred to in my original post referenced an
issue with the MD5 hash being different on either end due to differences in
the URI, causing a registration authentication problem (as I understand it).
I was just asking for assistance understanding what the link was recommended
as a fix.

Thanks!

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.484 / Virus Database: 269.12.1/963 - Release Date: 8/20/2007
5:44 PM

------------------------------

Message: 8
Date: Wed, 22 Aug 2007 08:53:28 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: [asterisk-users] rfc3680, reginfo+xml
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<442fbb120708212353u64235bffyfdba65518a20698f at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

RFC3680 defines a SIP event package for registration.
This event package which can be used through NOTIFY-SUBSCRIBE methods, seems
very useful for free sitting or presence applications.

This package is supported in various SIP phones (at least Thomson ST2030) :
when turned on, this feature adds a new login/logout menu among other
things.

It can also be used to send Welcome notices to mobile users : whenever a
mobile user comes in, a SIP MESSAGE is sent by a software application which
has previously subscribed to be notified of any registration event related
to this mobile user.

It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods.
But I couldn't find any trace of this specific Registration Event package
support (but I won't swear I searched the right way).

How can I make sure this feature is supported or not ?

More precisely, this Registration Event package support relies on these
headers :
SIP SUBSCRIBE "reg" Event
SIP SUBSCRIBE "application/reginfo+xml" Accept
SIP NOTIFY "reg" Event
SIP NOTIFY "application/reginfo+xml" Content

How shall I check ?

Regards
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Message: 9
Date: Wed, 22 Aug 2007 01:22:13 -0600
From: "Edgar Guadamuz" <eguadamuz at gmail.com>
Subject: [asterisk-users] How to re-read values from database in
	Trixbox
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<ffbf31780708220022p61011a07lf5ae62135774f2e5 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Hello guys,

I'm using Trixbox and I have a PHP application that updates a value in
the MySQL asterisk database as an interface to have a dynamic
customizable IVR.

After execute the UPDATE SQL query, the php application is supossed to
reload asterisk or restart amportal in order to get the change
working, but nor asterisk -rx reload nor amportal restart got the
change working.

So, the question is how can I re-read the new value from the database
to be effective in asterisk?

------------------------------

Message: 10
Date: Wed, 22 Aug 2007 10:42:52 +0300
From: Diego Iastrubni <diego.iastrubni at xorcom.com>
Subject: Re: [asterisk-users] How to re-read values from database in
	Trixbox
To: asterisk-users at lists.digium.com
Message-ID: <200708221042.53357.diego.iastrubni at xorcom.com>
Content-Type: text/plain;  charset="iso-8859-1"

You are updating the MySQL config, which is not propagated to the Asterisk 
config files. Only after you regenerate the configuratios, you can reload 
asterisk.

Dirty hack: "need_reload" flag must be set to true. 
Real solution: retrieve_conf + "asterisk reload"

On Wednesday 22 August 2007 10:22, Edgar Guadamuz wrote:
> Hello guys,
>
> I'm using Trixbox and I have a PHP application that updates a value in
> the MySQL asterisk database as an interface to have a dynamic
> customizable IVR.
>
> After execute the UPDATE SQL query, the php application is supossed to
> reload asterisk or restart amportal in order to get the change
> working, but nor asterisk -rx reload nor amportal restart got the
> change working.
>
> So, the question is how can I re-read the new value from the database
> to be effective in asterisk?

------------------------------

Message: 11
Date: Wed, 22 Aug 2007 20:05:38 +1200
From: Richard Scobie <r.scobie at clear.net.nz>
Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with
	Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <46CBEE52.9070607 at clear.net.nz>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Steve Totaro wrote:
> I guess I am just lucky to have 24 hour manned data centers with staff 
> that walk around looking for flashing LEDs.
> 
> I am sure there is some error thrown in /var/log/messages about a 
> failure that could be used to trigger a notification quite trivially.
> 

Both smartd and mdadm can be configured to send emails.

Regards,

Richard

------------------------------

Message: 12
Date: Wed, 22 Aug 2007 02:38:56 -0700 (PDT)
From: fateme fatah <faza_404 at yahoo.com>
Subject: [asterisk-users] How do I configure asterisk?
To: asterisk-users at lists.digium.com
Message-ID: <822230.64098.qm at web56609.mail.re3.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi: 
 Which one is better and easier for configure asterisk,directly or by GUI ? 
 I'd appreciate any idea. 
 Regards.

---------------------------------
Building a website is a piece of cake. 
Yahoo! Small Business gives you all the tools to get online.
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Message: 13
Date: Wed, 22 Aug 2007 02:40:10 -0700 (PDT)
From: fateme fatah <faza_404 at yahoo.com>
Subject: [asterisk-users] Which interface?
To: asterisk-users at lists.digium.com
Message-ID: <231245.55506.qm at web56611.mail.re3.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi: 
 If any body use meetmemanager or conman or web-meetme please say how about
is it.I'd appreciated any idea. 
 Regards.

---------------------------------
Be a better Heartthrob. Get better relationship answers from someone who
knows.
Yahoo! Answers - Check it out. 
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Message: 14
Date: Wed, 22 Aug 2007 06:23:36 -0400
From: "Raj Jain" <rj2807 at gmail.com>
Subject: Re: [asterisk-users] rfc3680, reginfo+xml
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<1971b0b60708220323xfd77a2fgf1622b2a507d7c50 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Olivier,

This feature is not supported in Asterisk. I can tell this looking at the
code.

If you want to test this yourself, send Asterisk a SUBSCRIBE message
with Event: reg header in it. You can either use an off-the-shelf UA
that supports RFC 3680 to do this or you can use SIPp (an open-source
SIP test tool) to do this. Since Asterisk does not support "reg"
event-package, it'll respond back with a 489 (Bad Event) response.

Raj

On 8/22/07, Olivier <oza-4h07 at myamail.com> wrote:
> Hi,
>
> RFC3680 defines a SIP event package for registration.
> This event package which can be used through NOTIFY-SUBSCRIBE methods,
seems
> very useful for free sitting or presence applications.
>
> This package is supported in various SIP phones (at least Thomson ST2030)
:
> when turned on, this feature adds a new login/logout menu among other
> things.
>
> It can also be used to send Welcome notices to mobile users : whenever a
> mobile user comes in, a SIP MESSAGE is sent by a software application
which
> has previously subscribed to be notified of any registration event related
> to this mobile user.
>
> It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods.
> But I couldn't find any trace of this specific Registration Event package
> support (but I won't swear I searched the right way).
>
> How can I make sure this feature is supported or not ?
>
> More precisely, this Registration Event package support relies on these
> headers :
> SIP SUBSCRIBE "reg" Event
> SIP SUBSCRIBE "application/reginfo+xml" Accept
> SIP NOTIFY "reg" Event
> SIP NOTIFY "application/reginfo+xml" Content
>
> How shall I check ?
>
> Regards
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

------------------------------

Message: 15
Date: Wed, 22 Aug 2007 12:26:07 +0100
From: "Adrian Marsh" <Adrian.Marsh at ubiquisys.com>
Subject: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<094A73044298734FB7D58CAAA319E1D663FDA5 at UBIQ-SERV1.ubiquisys.local>
Content-Type: text/plain;	charset="us-ascii"

Hi All,

A question for those with Cisco 7940/60 SIP phones.  I used to load
POS3-06-03-00 Firmware to the cisco phones.  A month or so ago, I ran
some tests and found that latest 3.8.6 firmware worked well, and solved
an issue or two on the phones.

I've a number of users who work outside of the LAN.  Our phones use DNS
names to talk to A*k, so in theory, just enabling NAT makes the phone
work outside the LAN (home users, remote users, etc).  However, when we
loaded the 3.8.6 firmware to these phones, we've found the phones no
longer work outside of the LAN.  Using Etherreal, we've found that the
communication between the Phone and A*k breaks (A*k never sees the
Register packets, but the phone does seem to send them.  I'll post more
detail if its needed, but I wondered if anyone else has seen this ? The
size of the IP packet for register is different (larger on the 3.8.6),
but the important content of the Register message seems the same.  I've
ruled out ISP/firewall interference, as its happened on a number of
users.

Obviously there are fixes in 3.8.6, so I don't want to downgrade the
phones again, but I can't see why they'd fail...

Adrian Marsh

------------------------------

Message: 16
Date: Wed, 22 Aug 2007 07:31:02 -0400
From: "Matt Florell" <astmattf at gmail.com>
Subject: Re: [asterisk-users] compatibility of PRI Two B channel
	transfers	TBTC/2BTC
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<61575c810708220431p7522cdder75de1ef09c16ec08 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On 8/21/07, Matthew Fredrickson <creslin at digium.com> wrote:
> Matt Florell wrote:
> > Hello,
> >
> > A client has asked for Two B channel Transfer capability (known as
> > TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG
> > Path Replacement) in a new Asterisk system and so I researched the
> > capability and came up with quite a few gaps in documentation.
> >
> > From what I've gathered, the official Digium statement is that is
> > works with DMS100 only, and only in Asterisk 1.4.X :
> > http://kb.digium.com/entry/26/140/
>
> This definitely works.  I wrote it and tested it myself.
>
> >
> > Although in a bugtracker posting with a patch from over two years ago,
> > Matt Fredrickson from Digium says that it works with 5ESS under
> > Asterisk 1.2.X:
> > http://bugs.digium.com/view.php?id=3554
>
> There's an implementation I scrubbed out a couple of years ago, but I
> think there was a bug in it that I was not able to fix.  When push came
> to shove, and I needed a switch to debug it on (and when I had more time
> to work on it), nobody offered switch access so that I could debug it.
> So I don't think it is working right now.
>
> > There are also bounties and claims of this feature working on NI2
> > protocol(although no patches posted) on the voip-info.org Wiki:
> >
http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+f
or+NI2+PRI+line
> >
http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20
channel%20transfer
>
> Yeah, well, they're really old :-)  Try getting a hold of the authors.

I am trying to, I have sent a message to whitehawk82 on the digium
forums and hopefully he will post back to me. If anyone knows who that
actually is, I would like to get a hold of them, Please email me their
contact details.

Thanks for clearing all of this up Matt, Hopefully I'll be able to fix
the notes out there to give a better picture of all of this once I'm
done with this project.

Thanks,

MATT---

------------------------------

Message: 17
Date: Wed, 22 Aug 2007 14:04:52 +0200
From: Lenz <lenz-ml at loway.it>
Subject: Re: [asterisk-users] DUNDi, So Easy A Caveman Could Do It!
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <op.txgmiefd3wzjep at pc-lenz>
Content-Type: text/plain; format=flowed; delsp=yes;
	charset=iso-8859-15

Well done! It's top-news on AstPligg right now.

http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_I
t

Thanks
l.

On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson  
<jmr.richardson at gmail.com> wrote:

> Here you go folks:
>
> ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
>
> If someone would be so kind as to upload to the wiki, it will be much
> appriciated.
>
> Thank you all who replied to my poll questions.
>
> As always, I hope this help.
>
> JR

-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.com

------------------------------

Message: 18
Date: Wed, 22 Aug 2007 14:13:20 +0200
From: Arnaud Ligot <arnaud at cblue.be>
Subject: Re: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <1187784800.6081.68.camel at CBlueLittle>
Content-Type: text/plain

FYI about cisco firmware:
http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml

A.

On Wed, 2007-08-22 at 12:26 +0100, Adrian Marsh wrote:
> Hi All,
> 
> A question for those with Cisco 7940/60 SIP phones.  I used to load
> POS3-06-03-00 Firmware to the cisco phones.  A month or so ago, I ran
> some tests and found that latest 3.8.6 firmware worked well, and solved
> an issue or two on the phones.
> 
> I've a number of users who work outside of the LAN.  Our phones use DNS
> names to talk to A*k, so in theory, just enabling NAT makes the phone
> work outside the LAN (home users, remote users, etc).  However, when we
> loaded the 3.8.6 firmware to these phones, we've found the phones no
> longer work outside of the LAN.  Using Etherreal, we've found that the
> communication between the Phone and A*k breaks (A*k never sees the
> Register packets, but the phone does seem to send them.  I'll post more
> detail if its needed, but I wondered if anyone else has seen this ? The
> size of the IP packet for register is different (larger on the 3.8.6),
> but the important content of the Register message seems the same.  I've
> ruled out ISP/firewall interference, as its happened on a number of
> users.
> 
> Obviously there are fixes in 3.8.6, so I don't want to downgrade the
> phones again, but I can't see why they'd fail...
>  
> Adrian Marsh
>  
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

------------------------------

Message: 19
Date: Wed, 22 Aug 2007 14:27:08 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: Re: [asterisk-users] rfc3680, reginfo+xml
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<442fbb120708220527u1e451382l98d97419d3d96aaa at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Thanks for replying, Raj.

Do you think such feature should, ideally, be implemented in Asterisk should
it be implemented in a dedicated software (presence ?) ?
It seems to me it should, though I'm not aware of many devices using this
feature, beside SIP hardphones.

Would it be difficult to extend current code to comply with this RFC, when
rfc3265 mechanism is already in place ?
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Message: 20
Date: Wed, 22 Aug 2007 05:32:58 -0700 (PDT)
From: satish patel <satish_patel_2000_2000 at yahoo.com>
Subject: [asterisk-users] asterisk with FAX problem
To: asterisk-users at lists.digium.com
Message-ID: <432665.73490.qm at web55515.mail.re4.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Dear all

                 I have setup of asterisk 1.2.14 and this is working fine.
first i want to explain you my setup of asterisk on network i have connect
my asterisk with mediant 2000 gateway and PRI terminated on mediant.

[fax_machin]------[audio_code_fxs]-----[Asterisk]-------[mediant_2000]---PRI
--<--<

  my fax machine connected with audiocode 24 fxs extention and which is
connected with asterisk and asterisk connected with mediant 2000 now i am
not able to send FAX outside my company so is there any special
configuration for T.38  protocal ?? can anyone explain me how do i go ahead
with this setup to start FAX

---------------------------------
Pinpoint customers who are looking for what you sell. 
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Message: 21
Date: Wed, 22 Aug 2007 22:32:59 +1000
From: "Klaverstyn, David C" <David.Klaverstyn at intergraph.com>
Subject: Re: [asterisk-users] Polycom and NAT
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<ED01933C8C6A2949A37045F012ACF7BABF51F4 at ap-mail1.ingrnet.com>
Content-Type: text/plain; charset="us-ascii"

I have both of those command lines for my natted sip device.

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darryl
Dunkin
Sent: Wednesday, 22 August 2007 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom and NAT

In your sip.conf, for the user:

nat=yes

To send keepalives for the UDP connection (depending on how flimsy the
device handling NAT is):

qualify=yes

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Klaverstyn, David C
Sent: Tuesday, August 21, 2007 17:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom and NAT

Hi All,

I have a Polycom 501 that is behind a NAT.  When it registers to the
Asterisk server it is using the IP address on the private network and
not the public IP of the NAT address.

Can someone tell me what I need to do so the phone registerers using an
internet address rather than the remote network NAT address.

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Message: 22
Date: Wed, 22 Aug 2007 15:33:12 +0300
From: Atis <atis at BEST.eu.org>
Subject: Re: [asterisk-users] How do I configure asterisk?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<945196e0708220533t22eed6eemc4074c78268206c4 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On 8/22/07, fateme fatah <faza_404 at yahoo.com> wrote:
> Hi:
>  Which one is better and easier for configure asterisk,directly or by GUI
?
>  I'd appreciate any idea.
>  Regards.

It's up to you to decide what's easier for you and your needs. For
beginners GUI is ok, but if you need some fancy functionality, you
will need to code config files for yourself. This question doesn't
have definite answer - some people prefer GUI management of their
servers, and thus choose MS IIS, and so on, but some prefer plain
config files (and choose Linux).

As a programmer i prefer config files (and that's more nerdy).

Regards,
Atis

-- 
Atis Lezdins,
IT Responsible of BEST Riga,
atis at BEST.eu.org
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

------------------------------

Message: 23
Date: Wed, 22 Aug 2007 08:08:37 -0500
From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
Subject: Re: [asterisk-users] Polycom behind NAT won't register to *
	server behind ALG
To: asterisk at voip-pbx.ca, 	Asterisk Users Mailing List -
	Non-Commercial Discussion	<asterisk-users at lists.digium.com>
Message-ID: <46CC3555.9040704 at fnords.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Henry L.Coleman wrote:
> I think what Alex was trying to say was that Polycom IP Phones are an
> example of immature product development. While they look very nice and
> have a nice display the product doesn't compete very well compared to
> other manufacturers.
> The two most obvious flaws are that they cannot be NAT'ed so they cannot
> be used as Off Premise eXtensions phones and the other being that they
> take so long to configure and re-boot. I have a golden rule with any phone
> that I plan on installing for a customer....If I can't get it working
> within 20 minutes then don't use it. I'm afraid Polycom breaks my golden
> rule.
> With such a lot of competition in this market they should have sorted this
> out two years ago.
> 

Reboots should not happen very often.  This is a non-issue for most people.

I've never seen a phone that could not work with NAT with Asterisk. 
Polycoms work just fine with NAT and Asterisk.  The nice thing about 
Asterisk's NAT support is that the phone does not need to support NAT.

------------------------------

Message: 24
Date: Wed, 22 Aug 2007 09:14:51 -0400
From: Russell Handorf <rhandorf at handorf.org>
Subject: Re: [asterisk-users] 99 bottles of beer
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <46CC36CB.1040704 at handorf.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

I've been working on an X10 component already. It works, but I wish the 
CMA15 would work correctly in Linux (I know it's suppose to, but for 
whatever reason the one I have just doesnt.) It's just a little AGI 
script that I have working with Cepstral that throws http PUTs to the 
Windows box that has Apache-PHP and the command line app. Yeah, I know. 
But it wouldnt be this tedious if the CMA15 would appear correctly on my 
* box.

(Oh, did I mention I made a LCARS Web GUI for this as well? :P)

Steve Edwards wrote:
> On Tue, 21 Aug 2007, Russell Bryant wrote:
> 
>> Nice!  While we're on the subject of silly but fun dialplan bits, check
out my
>> TV remote extension.  When I moved a few months ago, there was a while
when I
>> couldn't find the wireless keyboard that I usually use as my TV remote to
>> control MythTV.  So, I built dialplan so I could use a wireless phone as
my
>> remote, instead.  The dialplan reads digits from the phone and sends the
correct
>> commands to a MythTV network control interface for the frontend
application.
>>
>> I posted my tested .conf version and the untested AEL version to the
MythTV
>> wiki.  The AEL version would probably be prettier with macros, now that I
think
>> of it ...
>>
>>
http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using
_Asterisk
> 
> And practical :)
> 
> Almost every room in my house has a phone -- if I could teach my kids to 
> put them back where they belong.
> 
> This could easily be extended to recognize which phone was used so it 
> could control the Myth FE in that room.
> 
> Also, it could/should be extended to control x10 devices as well...
> 
> "To control the tv in this room, press 1. To control a tv in another room,

> press 2. To control the outside lights, press 3. To control the 
> sprinklers, press 4, ..."
> 
> Thanks in advance,
> ------------------------------------------------------------------------
> Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                             Fax: +1-760-731-3000
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

------------------------------

Message: 25
Date: Wed, 22 Aug 2007 09:50:29 -0400
From: "Steven" <asterisk at tescogroup.com>
Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with
	Asterisk
To: asterisk-users at lists.digium.com
Message-ID: <faheuq$o5n$1 at sea.gmane.org>
Content-Type: text/plain; charset="iso-8859-1"

For RAID1, I am not sure.

But for RAID 5, You should always use hardware RAID.

If you use software RAID and your CPU spikes for too long, you can corrupt
your disks. I have seen this several times.

-- 
-- 
Steven

http://www.glimasoutheast.org

  "Vidura Senadeera" <vidurased at gmail.com> wrote in message
news:afa14ef00708201903v209f8034lca006c8093251a2a at mail.gmail.com...

  Dear All,

  I would like to get community's feedback with regard to RAID1 ( Software
or Hardware) implementations with asterisk.

  This is my setup

  Motherboard with SATA RAID1 support
  CENT OS 4.4
  Asterisk 1.2.19
  Libpri/zaptel latest release
  2.8 Ghz Intel processor
  2 80 GB SATA Hard disks
  256 MB RAM
  digium PRI/E1 card

  Following are the concerns I am having

  I'm planing to put this asterisk server in production enviorment which is
having E1 connection to the asterisk server, approximately
  20 con-current calls, Music on hold, voice mail boxes.

  1. If I use Software RAID, what would be the impact to my deployment? (
problems that I have to face with regard to the call flow )
  2. If I use Hardware based RAID 1, what would be the impact to the system?
  3. According to your practical experiance what is the ideal solution among
both options?

  I will be highly appreciate your feedback on this regard.

  -- 
  Thanks & Regards,
  Vidura Senadeera,

----------------------------------------------------------------------------
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End of asterisk-users Digest, Vol 37, Issue 88
**********************************************