RE: [Asterisk-Users] CISCO POE

[Tarpo, Louie]

The POE switch needs to support always on.   Most switches check the device 
for 802.3af support before turning on power.   The phones only support the CDP 
power activation, not 802.3af.   I've used always on POE injectors from 
wireless access points successfully with Cisco phones.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of nik600
Sent: Wednesday, June 07, 2006 7:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CISCO POE


many thanks for your reply

i've tried to make  a cable with that configuration but it seems that
it doesn't work...

i'm using a 7905G Cisco ip phone and an ALL0484 Switch POE

thanks
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RE: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still workwith 8.2?

[Tarpo, Louie]

It does on my test phone.   Is your tftp server available?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brent
Torrenga
Sent: Monday, April 17, 2006 11:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still
workwith 8.2?


Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg
doesn't elicit any response from the phone using fw 8.2?


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com

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RE: [Asterisk-Users] Polycom 501

[Tarpo, Louie]

Both 
features work without AMP installed. Transfer works without any 
configuration changes. I've had both features working with bootrom 2.6.x 
and firmware 1.4.x - 1.6.x


 == the mailbox you want to subscribe to updates 
for (message waiting indicator), and  is your voicemail callback extension, 
and Z is the length of your extensions.


Polycom phone1.cfg (we use 
MAC-EXTEN.cfg):
msg 
msg.bypassInstantMessage="1" mwi 
msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" 
msg.mwi.1.callBack="" msg.mwi.2.subscribe="" 
msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" msg.mwi.3.subscribe="" 
msg.mwi.3.callBackMode="disabled" msg.mwi.3.callBack="" msg.mwi.4.subscribe="" 
msg.mwi.4.callBackMode="disabled" msg.mwi.4.callBack="" msg.mwi.5.subscribe="" 
msg.mwi.5.callBackMode="disabled" msg.mwi.5.callBack="" msg.mwi.6.subscribe="" 
msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/ 
/msg

Asterisk extensions.conf for 
1.0.x
exten 
= ,s,1,VoicemailMain(${CALLERIDNUM:-Z})
exten 
= ,s,2,Hangup



-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of 
AzfhasteriskSent: Thursday, March 02, 2006 7:13 AMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] Polycom 501

Try and download the 
correct sip.cfg for your boot ROM ver from here and see if it corrects the 
problem. We use AMP with these files and we never had an issue with the transfer 
button not working. 

http://www.freedomphones.net/polycom/files/

Make sure that you 
reboot the phone after you replace the sip.cfg and you should see the screen 
flash Updating config or something like that. It flashes quick so watch 
closely. If you dont see this then for some reason the phone is not connecting 
to the ftp/tftp server to get this config.

Just an FYI AMP does 
nothing with the phone sip configs 

Rick





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of MBIT TechnologiesSent: Thursday, March 02, 2006 5:03 
AMTo: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] Polycom 
501

AMP is being run but it 
seems the transfer needs to be configured in the phone somewhere so when you 
press the transfer button its like hitting #.


-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Mark AufflickSent: Thursday, 2 March 2006 10:51 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [Asterisk-Users] Polycom 
501

One thing 
to keep in mind when someone says "Asterisk does that by default" is that a lot 
of people have AMP installed, and an AMP installation includes extra 
configuration and features as well as the web interface. It may be that there is 
phone-specific config installed with AMP that is not installed in a base 
Asterisk installation. Cheers,Mark.-- Mark 
Aufflicke: [EMAIL PROTECTED]w: www.pumptheory.com (business)w: mark.aufflick.com (personal)p: +61 
438 700 647f: +61 2 9436 4737

On 
3/2/06, MBIT Technologies [EMAIL PROTECTED]  
wrote:


I guess it 
doesn't work by default on my phone. You still need to press hash to transfer 
calls. The transfer button doesn't work. Where do I set it?



Regards


Mark 
Brooker
T: 02 4959 
8670
M: 0415 846 
865
F: 02 9882 
0947
E: [EMAIL PROTECTED] 

W: http://www.mbit.com.au 



-Original 
Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Anton KrallSent: Thursday, 2 March 
2006 3:47 PMTo: 'Asterisk 
Users Mailing List - Non-Commercial 
Discussion'

Subject: RE: [Asterisk-Users] Polycom 
501


Those 
buyttons do work with asterisk by default... what kind of problems are you 
having?

  
  
  
  
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MBIT 
  TechnologiesSent: Wednesday, 
  March 01, 2006 7:56 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] Polycom 
  501
  Hi 
  Guys
  
  Just a quick question 
  regarding on the 501, has anyone been able to configure the transfer button 
  and messaging buttons to work with asterisk?
  
  Can you share a 
  configuration to do this?
  
  Thanks in 
  advance.






  
  

  iBurst Wireless Broadband from 
  $34.95/month - Platform Networks Spam  Virus Filtering by 
  Mail 
  SecurityTo 
  report SPAM forward the spam message to: [EMAIL PROTECTED]

  




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RE: [Asterisk-Users] What have I misconfigured?

[Tarpo, Louie]

Your voIpProt.server.1.expires= value in the Polycom sip.cfg is set to 
reregister with Asterisk every 60 seconds.  That's a bit much.  Most people use 
3600.   Also check your maxexpirey and defaultexpirey values in your asterisk 
sip.conf.

Louie

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jonathan k.
Creasy
Sent: Monday, September 12, 2005 2:59 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] What have I misconfigured?


I'm getting these messages every 7-10 seconds. 
-- Registered SIP '532' at x.x.x.x port 52956 expires 60
-- Registered SIP '532' at x.x.x.x port 56988 expires 60
-- Registered SIP '529' at x.x.x.x port 51444 expires 60
-- Registered SIP '529' at x.x.x.x port 64044 expires 60
-- Registered SIP '532' at x.x.x.x port 52956 expires 60
-- Registered SIP '532' at x.x.x.x port 56988 expires 60
-- Registered SIP '529' at x.x.x.x port 51444 expires 60
-- Registered SIP '529' at x.x.x.x port 64044 expires 60
-- Registered SIP '532' at x.x.x.x port 52956 expires 60
-- Registered SIP '532' at x.x.x.x port 56988 expires 60


532 started doing this last Thursday and 529 started doing it today. 

There are about 40 phones behind x.x.x.x.

The two phones in question are Polycom IP301's with 1 line setup on
them. 

There are 22 other Polycom IP301's there. 

-Jonathan
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RE: [Asterisk-Users] What have I misconfigured?

[Tarpo, Louie]

If they keep doing it, attach your Polycom config files and send them to the 
list.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jonathan k.
Creasy
Sent: Monday, September 12, 2005 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What have I misconfigured?


I'm hoping you guys might have some insight on that :) 

The first registration is filled out, the others are blank...it's only
registering one line

I'm going to locate those phones tomorrow and just restart them

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, September 12, 2005 8:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] What have I misconfigured?

Why is each phone registering twice (2 different ports)?

==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237   ICQ: 5662270
==


Jonathan k. Creasy wrote:
 I'll change thatI was thinking minutes, don't know why, it's
 always secondsthat still doesn't explain why these phones are
 registering every 14-20 seconds
 
 -Jonathan
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tarpo,
 Louie
 Sent: Monday, September 12, 2005 7:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] What have I misconfigured?
 
 Your voIpProt.server.1.expires= value in the Polycom sip.cfg is set to
 reregister with Asterisk every 60 seconds.  That's a bit much.  Most
 people use 3600.   Also check your maxexpirey and defaultexpirey
values
 in your asterisk sip.conf.
 
 Louie
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Jonathan
k.
 Creasy
 Sent: Monday, September 12, 2005 2:59 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: [Asterisk-Users] What have I misconfigured?
 
 
 I'm getting these messages every 7-10 seconds. 
 -- Registered SIP '532' at x.x.x.x port 52956 expires 60
 -- Registered SIP '532' at x.x.x.x port 56988 expires 60
 -- Registered SIP '529' at x.x.x.x port 51444 expires 60
 -- Registered SIP '529' at x.x.x.x port 64044 expires 60
 -- Registered SIP '532' at x.x.x.x port 52956 expires 60
 -- Registered SIP '532' at x.x.x.x port 56988 expires 60
 -- Registered SIP '529' at x.x.x.x port 51444 expires 60
 -- Registered SIP '529' at x.x.x.x port 64044 expires 60
 -- Registered SIP '532' at x.x.x.x port 52956 expires 60
 -- Registered SIP '532' at x.x.x.x port 56988 expires 60
 
 
 532 started doing this last Thursday and 529 started doing it today. 
 
 There are about 40 phones behind x.x.x.x.
 
 The two phones in question are Polycom IP301's with 1 line setup on
 them. 
 
 There are 22 other Polycom IP301's there. 
 
 -Jonathan
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RE: [Asterisk-Users] Good Polycom Dealer?

[Tarpo, Louie]

We use both voipsupply and tritechcoa.  We've had no problems with either one.  
I've received firmware from each company.

Louie

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny Kant
Sent: Tuesday, September 06, 2005 3:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Good Polycom Dealer?


Could any of you provide me information on a good
Polycom phone dealers to utilize.  One who provides
firmwares ..etc 


Thank you!

Kenny 






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RE: [Asterisk-Users] Good Polycom Dealer?

[Tarpo, Louie]

I haven't had the pleasure of dealing with tritech's RMA department.  So I 
can't speak on that aspect of their business.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Welter
Sent: Tuesday, September 06, 2005 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Good Polycom Dealer?


Maybe you should read tritechcoa's return policies before you start 
recommending them.  After suffering through their RMA juggernaut, I'll 
never again do business with them.




Tarpo, Louie wrote:
 We use both voipsupply and tritechcoa.  We've had no problems with either 
 one.  I've received firmware from each company.
 
 Louie
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny Kant
 Sent: Tuesday, September 06, 2005 3:43 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Good Polycom Dealer?
 
 
 Could any of you provide me information on a good
 Polycom phone dealers to utilize.  One who provides
 firmwares ..etc 
 
 
 Thank you!
 
 Kenny 
 
 
 
 
   
   
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RE: [Asterisk-Users] Cisco 7960 / SIP tftp configs

[Tarpo, Louie]

I've not had the deleting lines problem.  When you're deleting a line, are you 
changing the config to ... 
line6_name: 
line6_displayname: 
line6_shortname: 
line6_authname: 
line6_password: 


#Change lineX_shortname:  to whatever you want them to see on the LCD.
line4_name: 
line4_displayname: 
line4_shortname: Line4
line4_authname: 
line4_password: password



#Change  to your VoiceMailMain() extension
messages_uri: 


Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk
User Group
Sent: Wednesday, August 24, 2005 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 / SIP  tftp configs


I have three questions about my 7960 phone that I can't discern from the 
docs/wiki.

1st - If I change the SIPxx.cnf file to change registrations it sets 
up new lines as expected. If I delete a line it doesn't get removed when 
I reboot the phone. I have to go to the phone, unlock it, and reset the 
SIP parameters. How do I make it forget what it has programmed and 
listen only to the download?

2nd - Has anyone figured out how to get the Message button to launch a 
dial to VoicemailMain?

3rd - How do I display on the LCD an alias to the registered line?
line1_name: 2000
line1_authname: 2000
line1_password: **

The doc seems to suggest that line1_name is what it registers with and 
line1_authname is what it uses if challenged during the 
authentication. This doesn't make any sense to me. I am looking for the 
line to be 2000 but the display to say Home or Business, etc.

Thanks, dbc.
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RE: [Asterisk-Users] Can not dial more then 23 calls

[Tarpo, Louie]

It looks like you are sending calls out over one port.  To help you out, we 
will need to look at your extensions.conf and zapata.conf.  My hunch is that 
you are dialing out using something like 
Dial(zap/g3/${EXTEN},20,) where the group of channels you're using is on one 
port of your Digium card.

If my math is right, you should be able to send 69 calls long distance, and 23 
local calls at a time with no failover.

Louie Tarpo
Adam Aircraft

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Pudenz,
Duane 
Sent: Wednesday, August 17, 2005 12:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can not dial more then 23 calls


We are testing our Asterisk server prior to deployment.  The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.

We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.

We can not get more then 23 total active calls to connect to the test
numbers, the test numbers terminate to another PBX that we can monitor.
We have dialed out using cell phones to this other PBX while the test is
happening and it connects, meaning it has more then 23 active calls on
it.

If we place more then 23 calls then it seems to 'queue' the extra calls,
though not all of the extra calls complete after we stop adding new
calls.  They seem to get stuck in a queue or lost.  We will send 200
calls through the Asterisk server and all but about 20 do eventually
complete.  Those 20 or so are stuck as Asterisk thinks the channels are
busy with the calls when in fact there are no 'real' calls on the
server.

We can send 30 calls through the LD or PRI and only 23 are actually
connected at a time.  We can send 30 calls to both LD and PRI at the
same time and still only a mixture of 23 calls are actually active at
one time.

So our issue seems to be located in our Asterisk server.  Is there a way
to limit or throttle an Asterisk server so that it will not place more
then 'x' calls?  

We need to be able to support 48 calls.

Any ideas?

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RE: [Asterisk-Users] 8 FXS in Asterisk Server

[Tarpo, Louie]

As would I. we've got 4 SPA-3000s in one location that aren't quite getting the 
job done.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sean Rima
Sent: Monday, August 15, 2005 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 8 FXS in Asterisk Server


Joseph wrote:
 Easy and cheap.
 Get two gateways AG-468 (each have 4  FXS ports) made by Atcom
 http://www.voip-info.org/tiki-index.php?page=Atcom
 
 one is about 88/ea
 I have two on the way and will let you know how it works.
 

I would be interested in knowing how these work as well

Sean

-- 
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Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie
FreeWorldDial 689482
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RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone

[Tarpo, Louie]

Which is where the dial 9 for an outside line came from.  Just make sure no 
internal extensions start with 9.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tom Rymes
Sent: Thursday, August 11, 2005 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone


The difficulty is making the phone dial quickly when you dial a three  
or four digit extension number, yet not having it dial so quickly  
that it screws up a user who dials the first four digits of a 10  
digit number and has to look down at a piece of paper to read the  
last 6 digits. It's pretty annoying when that happens and the phone  
has already initiated the call. (I would make the phone wait either  
one or two seconds to match an internal extension.)

Tom

On Aug 11, 2005, at 5:11 PM, Tarpo, Louie wrote:

 You write out a dialplan, then when you match a pattern in the dial  
 plan, the Polycom will initiate the call immediately.  This way you  
 can have 4 digit internal extensions dial immediately, or have it  
 wait for a long distance or international number.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Andrew M
 Stemen
 Sent: Thursday, August 11, 2005 3:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
 systemtoreplace an old PBX but using existing phone


 Jonathan k. Creasy wrote:

 YeahI think that every install I have done the first thing that
 happens is why is there a delay before the call connects? and the
 answer is you have to hit dial or wait 10 seconds.


 What all phones does that apply to? I'm fairly certain it applies  
 to the
 Polycom phones I've read about, but I'm not sure about others. I'm
 obviously a newbie to the field as well (well, at least to the  
 physical
 phones).
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RE: [Asterisk-Users] Newbie Question: Building an Asterisk systemtoreplace an old PBX but using existing phone

[Tarpo, Louie]

You write out a dialplan, then when you match a pattern in the dial plan, the 
Polycom will initiate the call immediately.  This way you can have 4 digit 
internal extensions dial immediately, or have it wait for a long distance or 
international number.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew M
Stemen
Sent: Thursday, August 11, 2005 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
systemtoreplace an old PBX but using existing phone


Jonathan k. Creasy wrote:
 YeahI think that every install I have done the first thing that
 happens is why is there a delay before the call connects? and the
 answer is you have to hit dial or wait 10 seconds. 

What all phones does that apply to? I'm fairly certain it applies to the 
Polycom phones I've read about, but I'm not sure about others. I'm 
obviously a newbie to the field as well (well, at least to the physical 
phones).
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RE: [Asterisk-Users] Help me how to listen voicemail with SIP 7960

[Tarpo, Louie]

In your SIPmac.cnf file for your Cisco phones, change the line messages_uri: 
3399 to whatever extension your VoiceMailMain() is set to on your Asterisk 
box.  In our case, it's 3399.  Also, there is newer firmware for the phones, 
P0S3-07-5-00.

Louie

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lokesh
kumar
Sent: Wednesday, August 10, 2005 12:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help me how to listen voicemail with SIP 7960


Hi, 
Everybody

I am running asterisk successfully, I am having few
couples of SIP 7960 phones, I am booting the phones
with P0S-3-06-0-00 file. But i am unable to access
voicemails through the phone, but i can send voicemail
attachments with the email, which i mentioned in
voicemail.conf file.
The messages button never respond when i press it,
suggest me how i have to access voicemail boxes
through SIP 7960 phones.
I will be very thank full to you

Lokesh
Portugal
mail - [EMAIL PROTECTED]







Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to 
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RE: [Asterisk-Users] DTMF issues with SIPPhone?

[Tarpo, Louie]

We encountered the same issue.  change dtmfmode=rfc2833 to dtmfmode=inband and 
make sure you're using a ulaw connection.  If you use a lossy codec, it will 
scramble the DTMF tones.


Your config would change like so,

[sipphone]
type=peer
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
fromuser=1747xxx
username=1747xxx
password=x
context=fromsipphone
dtmfmode=inband
canreinvite=no
disallow=all
allow=ulaw

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 10:27 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] DTMF issues with SIPPhone?


Does anyone else have DTMF issues with SIPPhone?  When calling into my
DID, and entering, say, 1002.  Sometimes it will recognize it properly
(rarely), other times it will receive something different.  Such as,
1102 or 1000, etc.  Has anyone else been having these issues?  I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:

[sipphone]
type=peer
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
fromuser=1747xxx
username=1747xxx
password=x
context=fromsipphone
dtmfmode=rfc2833
canreinvite=no


Any ideas?  Am I doing something wrong?

Thanks!
-JD-
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RE: [Asterisk-Users] DTMF issues with SIPPhone?

[Tarpo, Louie]

Yes we are.  I just double checked our line, and oddly, the dtmf tones aren't 
getting sent to our asterisk server.  Switched it back to rfc2833, and it 
works.  It was the other way around when I first connected us.  Some informal 
testing just now doesn't show the DTMF tone problem in rfc2833 mode that you're 
having.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone?


Louie,

On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
 We encountered the same issue.  change dtmfmode=rfc2833 to dtmfmode=inband 
 and make sure you're using a ulaw connection.  If you use a lossy codec, it 
 will scramble the DTMF tones.

Are you using SIPPhone?  When I use dtmfmode=inband, it just doesn't
recognize the tones at all..
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RE: [Asterisk-Users] DTMF issues with SIPPhone?

[Tarpo, Louie]

RFC2833 is sent out of band.  What's the output on your asterisk console?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone?


So the way I understand this is with rfc2833, DTMF is sent out of
band.  So does this mean that SIPPhone is interpreting the tones
incorrectly?  Asterisk shouldn't be doing any actual tone detection
with this method, right?

On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
 Yes we are.  I just double checked our line, and oddly, the dtmf tones aren't 
 getting sent to our asterisk server.  Switched it back to rfc2833, and it 
 works.  It was the other way around when I first connected us.  Some informal 
 testing just now doesn't show the DTMF tone problem in rfc2833 mode that 
 you're having.
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RE: [Asterisk-Users] DTMF issues with SIPPhone?

[Tarpo, Louie]

I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for 
my purposes.  I've been really pounding on our sipphone number the past half 
hour or so and I'm seeing the same issues you are.   Sometimes it hits 
correctly, sometimes it doesn't.   IE, Dialing 5954, some of the times it 
works, sometimes it's 5594, 5994, 5944, 59, 95, 54, etc.   I know I'm not 
fatfingering the dialing because my cell prints the dtmf digits to the screen.  
We haven't been seeing the issues here because our sipphone number isn't 
published (yet).

Louie



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone?


On 8/8/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
 RFC2833 is sent out of band.  What's the output on your asterisk console?

I don't see any output during this time on my asterisk console. 
Unless there's additional logging I'd need to enable?

Thanks for the help!
-JD-
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

We do extension by extension is our dialing plan because we have a wildcard at 
the end trapping all unused extensions and playing a this extension is not in 
use message and forwarding users into our IVR.  It depends on individual 
circumstances which works better.  We have 300 DIDs for our sip phones, and 
only 50 in use.  Those 50 are also not sequential extensions.  So it's less 
painful to approach this way for our circumstance.  If you had all of your 
extensions in use, the wildcard would be easier and cleaner.  Then if you 
needed to remove one, include a [not-in-service] context above the in use 
extensions.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


But why do it that way?

Wouldn't:

exten = _72X,1,Dial(SIP/${EXTEN},50)

Be ideal? Or at least an easier way to expand the dialplan without mucho
administration?

Just a question...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

They both work.  Without a wildcard in that, you don't really gain anything by 
doing ${EXTEN} in that case except for changing the one number during 
copy/paste operations.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of jj
Sent: Thursday, August 04, 2005 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question


Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

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RE: [Asterisk-Users] Polycom phones w/ two lines on different servers

[Tarpo, Louie]

My Polycom 300 is registered on two different servers on two different subnets. 
  It was failing the same way for me as well because we had server information 
in sip.conf, so it was always going to one server.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Robins
Sent: Thursday, August 04, 2005 2:41 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Polycom phones w/ two lines on different
servers


This is in the -app.log file:

0804194926|sip  |4|00|Registration failed User: 1800, Error Code:403
Forbidden

Where '1800' is the extension I am attempting to register.  SIP.conf is
set up properly, and there is nothing in Asterisk showing a denied
registration attempt.

Could it be because the second server is on a different subnet across a
WAN link?  There is no firewall between the phone and the servers.


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Thursday, August 04, 2005 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different
servers

I use the default. Try this.
cd /home/PlcmSpIP
cat log/YOURMAC-boot.log

se what the log file says, also do the same with the YOURMAC-app.log


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

We handled it by creating a macro which dials the exten, then sends the call to 
voicemail.  

You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc

or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})

then the macro would look something like 
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)

exten = s-NOANSWER,1,VoiceMail(u${ARG1})   ;Send 
to voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup

exten = s-BUSY,1,VoiceMail(b${ARG1})   ;Send to 
voicemail, play busy message
exten = s-BUSY,2,Hangup

exten = _s-.,1,Goto(s-NOANSWER,1)

Depends on your needs which way would work better.  We define extension by 
extension individually, then have a wildcard at the end that plays a message 
that says the extension is not in use and then puts them in our main menu.  In 
case we have to remove or change an extension individually.

Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question


I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server.  I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc 

in my from-sip context I began doing entries such as:


exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)


exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)


..etc ..etc

This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck.  Is there anyway around
this?

Thanks!

Kenny





Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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RE: [Asterisk-Users] newbiew extensions.conf question

[Tarpo, Louie]

He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.

You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})

and so forth.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question


That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.

I don't follow your logic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question

Right track, but it can be simplified even more

exten = 720,1,macro(sipexten,${EXTEN})


On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:

 We handled it by creating a macro which dials the exten, then sends  
 the call to voicemail.

 You could create it where each extension is handled seperately
 exten = 720,1,Macro(sipexten,720)
 exten = 721,1,Macro(sipexten,720)
 etc

 or you could handle them all in a group with wildcards
 exten = _72x,1,Macro(sipexten,${EXTEN})

 then the macro would look something like
 [macro-sipexten]
 exten = s,1,NoOp(${CallerIDNum})
 exten = s,2,Dial(SIP/${ARG1},24)
 exten = s,3,Goto(s-${DIALSTATUS}, 1)

 exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to  
 voicemail, play unavailable message
 exten = s-NOANSWER,2,Hangup

 exten = s-BUSY,1,VoiceMail(b${ARG1});Send to  
 voicemail, play busy message
 exten = s-BUSY,2,Hangup

 exten = _s-.,1,Goto(s-NOANSWER,1)

 Depends on your needs which way would work better.  We define  
 extension by extension individually, then have a wildcard at the  
 end that plays a message that says the extension is not in use and  
 then puts them in our main menu.  In case we have to remove or  
 change an extension individually.

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Kenny  
 Kant
 Sent: Thursday, August 04, 2005 4:07 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] newbiew extensions.conf question


 I am newbie trying to setup about 12 Polycom Ip500's
 on an asterisk server.  I am working on my
 extensions.conf and am trying to make it so that all
 my extensions can dial each other. My extensions are
 number 720, 721, 722, 723 ..etc

 in my from-sip context I began doing entries such as:


 exten = 720,1,Dial(SIP/720,20)
 exten = 720,2,Voicemail(u720)


 exten = 721,1,Dial(SIP/721,20)
 exten = 721,2,Voicemail(u721)


 ..etc ..etc

 This is not a big deal for such a small number of
 extensions but I was thinking about larger installs..
 this would begin to suck.  Is there anyway around
 this?

 Thanks!

 Kenny




 
 Start your day with Yahoo! - make it your home page
 http://www.yahoo.com/r/hs

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RE: [Asterisk-Users] Some more VOICEMAILMAIN issue...

[Tarpo, Louie]

You 
need to match the extensions you have in voicemail.conf to the callerid you're 
passing to voicemailmain(). For5 digit extensions it would be 
exten = 22999,1,VoiceMailMain(s${CALLERIDNUM:-5)


Louie

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Mauro 
  ZaninSent: Tuesday, July 26, 2005 12:56 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Some more 
  VOICEMAILMAIN issue...
  Hi everybody,
  I have corrected this line in extensions.conf by 
  stripping spaces off and now it executes:
  
  
  exten = 
  22999,1,VoiceMailMain(s${CALLERIDNUM})
  when it runs, the mail box number is asked and 
  password too. I expected no question were made, because I inserted 
  CALLERIDNUMBER and s in front of box number.
  Anybody knows why?
  Thank to you all, very kind members of this 
  list!
  Ciao
  Mauro
  
  
  
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RE: [Asterisk-Users] IP Phone with Standard Power Ethernet

[Tarpo, Louie]

Half of my 7960G phones work with standard POE, the other half work with the 
special rewiring.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin P.
Fleming
Sent: Tuesday, July 12, 2005 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP Phone with Standard Power Ethernet


Michael J. Tubby B.Sc (Hons) G8TIC wrote:

 Cisco 7940 and 7960 phones without the G (global) suffix used Cisco 
 PoE and had their keys labled in local languages. The 7940G and 7960G 
 Global phones are IEEE 802.3af PoE and have the keys engraved with 
 icons and stick over labels.

Are you sure about that? I have users with 7960G phones (icon buttons) 
that did not work without making cross-over patch cables from our PoE 
injectors.
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RE: [Asterisk-Users] DELL 2800 : PCI Parity error

[Tarpo, Louie]

We experienced the same problem on a Dell 2850 server.  Our other asterisk 
admin went a different route and inquired with Dell.   They told him this was 
completely normal and not to worry about it.  I'm still skeptical.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Syed Akbar
Sent: Sunday, July 10, 2005 7:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DELL 2800 : PCI Parity error


I had the same problem with a Dell PowerEdge 800 server and a TDM400 card. I
talked to Digium and they suggested a workaround by adding a NMI flag reset
in the Linux boot file. This only prevents a system lockup. The system
worked fine even with the blinking orange light and the dazed and confused
comment from the modprobe command. I have heard that the new firmware on the
TDM400P card has fixed this problem, but have not experienced that first
hand. 

In the same machine I am using a new TE110P with no problems at all.

Syed Akbar

Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Sunday, July 10, 2005 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DELL 2800 : PCI Parity error

I too had this problem, on a 2850, as well as the occasional missed IRQ.

I went through all the usual zaptel tuning stuff Disabled fb, disabled
ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel 
card to second CPU so all interrupts from zaptel are on their own. My
systems now run close to 100% in zttest, never miss an irq and don't seem to

generate PCI parity errors any more.

I don't know if I've fixed it, but you should really go through the whole
process anyway.


==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237   ICQ: 5662270
==


list wrote:
 Still not resolved
 
 On Wed, 2005-06-08 at 01:16, David John Walsh wrote:
 
Frank

Did you ever resolve this?  If so what was the issue?

On 03/05/05, list [EMAIL PROTECTED] wrote:

Hi,
I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error
(EB113 on the display)
I am learning linux and asterisk as I go along, there might be obvious
things I should know, but bear with me.

From demsg below my 2 digium cards installed are listed (no config or
connections done to digium cards yet), the conflict is with the TDM400P
card, without that card, in any slot, no alarm.

Zapata Telephony Interface Registered on major 196
Registered Tormenta2 PCI
Controller version: 24
FALC version: 
TE110P: Setting up global serial parameters for E1 FALC V1.2
TE110P: Successfully initialized serial bus for card
Found a Wildcard: Digium Wildcard TE110P T1/E1
Freshmaker version: 71
Freshmaker passed register test
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
Registered tone zone 8 (Norway)
TE110P: Span configured for CCS/HDB3/CRC4
Calling startup (flags is 4099)
wcte1xxp: Setting yellow alarm
usb.c: registered new driver wcusb
Wildcard USB FXS Interface driver registered
TE110P: Span configured for CCS/HDB3/CRC4
Calling startup (flags is 4099)
Registered tone zone 8 (Norway)
TE110P: Span configured for CCS/HDB3/CRC4
Calling startup (flags is 4099)
Registered tone zone 8 (Norway)

ramchip problem is false, without the card all ok, ramtests on machine
as well.

lsmod shows wcusb driver on zaptel, I dont need that, can I remove it?
is that a problem or not?

# lsmod
Module  Size  Used byNot tainted
usbserial  23964   0  (autoclean) (unused)
lp  9156   0  (autoclean)
parport38848   0  (autoclean) [lp]
autofs416984   0  (autoclean) (unused)
wcusb  19552   0  (unused)
wctdm  41088   0  (unused)
wcte11xp   22048   0  (unused)
zaptel182080   4  [wcusb wctdm wcte11xp]
e1000  77884   1  (autoclean)
floppy 57552   0  (autoclean)
sg 37388   0  (autoclean)
microcode   6912   0  (autoclean)
ide-cd 34016   0  (autoclean)
  

RE: [Asterisk-Users] SIP Phone Config Generator

[Tarpo, Louie]

We use a mix of Cisco/Polycom phones. I just keep generic template files and 
make a copy of them for each new phone.  It would be fairly trivial to put a 
webserver on the same server with the tftpd.  In php or perl it would be fairly 
trivial to make a webpage that copied template files, replaced default values, 
and wrote out SIPmac.cnf or mac.cfg and mac-ext.cfg.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Max Clark
Sent: Tuesday, June 28, 2005 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Phone Config Generator


Hi all,

Cisco/Polycom phones will pull their configuration via a tftp server to 
help manage mass deployments of phone systems. Are there any config 
generators available that will create the file for the tftp server?

TIA,
Max
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RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems

[Tarpo, Louie]

That is not a typo.  One is the loader, the other is the firmware.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Geoff
Manning
Sent: Thursday, June 23, 2005 6:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems


 I have a second-hand 7960 which I am attempting to upgrade to 
 use a SIP
 image.
 
 The phone currently has a firmware release which doesn't seem 
 to be listed
 in Cisco docs - P003AM30. On reboot, it finds the tftp server 

Here's how I performed the upgrade:

Downgrade from the stock P003AM30 to POS30203

Upgrade to version 5.1 (first signed binary firmware)

Upgrade to version 7.1 * (most recent version? maybe 7.4?)

* When upgrading to 7.1 there is a typo in the OS79XX file, it will say
P00x change it to P0Sxgreat typo by Cisco.


Check the comments on this wiki page: 

http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx?page=Asterisk%20phon
e%20cisco%2079xxcomments_threshold=0comments_offset=0comments_sort_mode=c
ommentDate_desccomments_maxComments=10comments_parentId=353#threadId358

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RE: [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems

[Tarpo, Louie]

If you've followed the advice of previous posters, delete your 7-4 firmware and 
re-extract it.  You should NOT rename it.

Your OS79XX.txt should contain
P003-07-3-00

and your SIPMAC.cnf should contain
# SIP Configuration Generic File (start)
image_version: P0S3-07-4-00 

The P003-07-3-00 is the loader file
the loader file loads the actual image which is the P0S3-07-4-00

Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Patrick
Lidstone (Personal E-mail)
Sent: Thursday, June 23, 2005 3:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems



 Make sure that you have done the following:

 1.) Set up the phone to use DHCP to get an address *or* manually  
 configured an e-mail address using the settings on the phone.

 2.) Set the DHCP server to give out the correct TFTP server address,  
 *or* configure Alternate TFTP Server = yes and manually specify the  
 server address.

 I assume that you have already done that, but you never know!

 Tom

Hi Tom, the problem is that the Universal Application Loader has never
successfully loaded a firmware image, so there is no way to set these
options manually. The phone is definitely doing something with DHCP, but
never generates a TFTP request - apparently?

Patrick

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RE: [Asterisk-Users] Includes include the includes?

[Tarpo, Louie]

If you set up only internal extesions in the default context, then include 
default in [building1] and [office] all of those extensions can call 
internally.  I set up several standard features into the default context which 
everyone can access.

If you want to control feature access, say, for example a line that reads the 
time (let's just say), put that in a different context.

[office]
include = default
include = local
include = international
include = timeline

[building1]
include = default
include = local
 
[default]
exten = 700,1,Dial(SIP/${EXTEN})
exten = 100,1,Dial(SIP/${EXTEN})
exten = 200,2,Dial(SIP/${EXTEN})
;and so on for your other extensions

[timeline]
exten = something

Or, if you wanted to control access to who could call which internal extension, 
then you break out default into groups of their own.

[office]
include = office-ext
include = local
include = international
include = timeline

[building1]
include = building1-ext
include = office-ext
include = local

[office-ext]
exten = 700,1,Dial(SIP/${EXTEN})

[building1-ext]
exten = 100,1,Dial(SIP/${EXTEN})

In that example, office can call other office numbers, make local calls, make 
international calls, and access the timeline feature.  Building 1 can access 
building1 extensions, office extensions, and make local calls.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Mason
(Lists)
Sent: Friday, June 17, 2005 5:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Includes include the includes?


First, let me apologize for the multiple posts - my procmail recipe had a
bug that hid most mail form the list for a day.

The inheritance of includes creates a problem for me. I want to group the
extensions, not put them all in default to control access to features. So
[office] extensions should have the include = longdistance but  [building1]
should not.

However, how can [building1] then dial office?


Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759  

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tarpo, Louie
 Sent: Wednesday, June 15, 2005 9:03 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Includes include the includes?
 
 Yes it does.  You want something like this...
 
 [office]
 include = default
 include = local
 include = international
 
 [building1]
 include = default
 include = local
 
 [default]
 exten = 700,1,Dial(SIP/${EXTEN})
 exten = 100,1,Dial(SIP/${EXTEN})
 exten = 200,2,Dial(SIP/${EXTEN})
 ;and so on for your other extensions
 

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RE: [Asterisk-Users] Sipura 3000 help

[Tarpo, Louie]

There is a better way.  Upgrade to the latest 3.1.3a firmware, which supports 
PSTN to VoIP gateway calls.  Specifically the Off hook while calling VoIP 
option needs to be no.

If you need help configuring it, use the SPA3K w/Asterisk configurator at 
Voxilla http://voxilla.com/spa3kasterisk.php is adequate.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Thursday, June 16, 2005 4:42 PM
To: Adrian A; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura 3000 help



 Anyone know what I need to do to get the FXO port on the SPA 3000 to
 forward calls to Asterisk?  My Asterisk is running on port 5061 and I
 set the dial plan on the device to forward to [EMAIL PROTECTED]:5061 but
 Asterisk is not picking it up.  I can see on tcpdump traces that the
 Invite packets do go to through to the asterisk machine on port 5061,
 but it's not picking them up.  sip debug does not show any packets
 either.

Take a look at this:
 http://lists.digium.com/pipermail/asterisk-users/2005-March/097922.html


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RE: [Asterisk-Users] HT-488 vs. SPA-3000?

[Tarpo, Louie]

We have 6 SPA3000s.  The device is extremely configurable and works 
inbound/outbound with Asterisk with the latest firmware update with little 
trouble.  However, we've yet to resolve sound volume and quality issues.  The 
PSTN to SPA gain and SPA to PSTN gain along with FXS Port Input Gain and Output 
Gain settings have had no positive effect.  The problem is entirely with the 
analog line adapter.  VoIP calls from the analog phone to other VoIP 
destinations are perfect.  We also have several SPA-1001s and SPA-2000s that 
have been running perfect since day 1.

Also Sipura support is nonexistant.  Just our experience.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Littlejohn
Sent: Wednesday, June 15, 2005 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000?


I have only had experience with the Sipura 3000 and I would agree with
the voice volume problems.  I have given up on it working properly
(adjusted gains, impedences, firmware, etc), the voice quality is just
to low to actually use.  I actually purchased a second one thinking
that the first might be defective.

Would not recommend it because of the low sound volume problem. 
Talking on the phone is actually the point of the device so who cares
how configurable it is if you cannot hear anything.  I purchased a
Digium TDM400P and have had very good luck with it.

Dan

On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote:
  Just want to tap the collective wisdom of this list as to experiences
  pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
 
 I've not played with the ht488, but I believe others have posted this
 device does not provide access to the pstn-fxo port. The spa3k does
 provide that access (if you want it).
 
  Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
  the top of the pick..Any comments and experiences esp. with Asterisk
  compatibility would be great, before I plonk in the bucks.
 
 The spa3k works fine with asterisk as many have posted. However, once
 in awhile it does act a little strange in two different ways:
  1. the spa3k will sometimes interpret some voices as tones which cause
  a little disturbance to any conversation going on. It is sort of like
  the old telephony talk off that existed years ago. Doesn't happen
  all that often and seems to be more sensitive to female voices based
  on my one-year of experience.
  2. sometimes it seems to operate in half-duplex mode, where if you try
  to talk at the same time as the other end is talking, the other end
  won't hear you.
 
 Neither one of those have been all that objectionable to me, but they
 happen and others have posted roughly the same issues. I've not heard
 of anyone that has found a way to minimize those two issues.
 
 The down side of the spa3k right now is that Cisco bought the company
 and there likely won't be much advancement of the code until after the
 ownership (and development efforts) are sorted out by both companies.
 (The same kind of product delays has been seen with their Linksys
 purchase, as well as when other companies are bought/sold.)
 
 Its fairly common knowledge that ex-Cisco folks started Sipura for the
 sole purpose of selling the company for a hugh profit. Their success
 in accomplishing that objective could only be measured in terms of
 producing Sipura products that had at least some acceptance of those
 products by end users. With those previous objectives accomplished,
 how will Cisco handle the Sipura products in the future? (It's any-
 one's guess at this point since Cisco also has at least some track
 record of mismanaging purchased companies for whatever reason.)
 
 From an internal Cisco strategic perspective, they now own the assets
 that can make a major dent in the mass-market end-user voip product
 arena, and hopefully they'll take that in a positive direction.
 
 Given the price of the spa3k, I don't have any issue with purchasing
 more of them right now. Excellent choice for the one-to-three pstn-fxo
 market space.
 
 
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RE: [Asterisk-Users] HT-488 vs. SPA-3000?

[Tarpo, Louie]

I'm curious what other standalone FXO adapters work with Asterisk.  At 
everything from the default to the maximum in positive and negative values, and 
combination of gain settings, we still get unacceptable distortion and echo.  
I've checked the phone lines, they work normally with a regular phone.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Wednesday, June 15, 2005 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] HT-488 vs. SPA-3000?


The majority of the audio level issues seem to be on the fxo port
and setting the transmission levels (gain) to compensate for the 
cable loss to the central office. Eg, setting the pstn gain values
to what should be appropriate causes echo, etc, not unlike the TDM
card. (I have both in use.)

In other words, the further the spa3000 (or TDM card) is from the
central office, the more difficult it seems to be to set gain values
that are acceptable. That's apparently why many people find its use
is okay while others seem to think its objectionable.


 We have 6 SPA3000s.  The device is extremely configurable and works 
 inbound/outbound with 
Asterisk with the latest firmware update with little trouble.  However, we've 
yet to resolve 
sound volume and quality issues.  The PSTN to SPA gain and SPA to PSTN gain 
along with FXS Port 
Input Gain and Output Gain settings have had no positive effect.  The problem 
is entirely with 
the analog line adapter.  VoIP calls from the analog phone to other VoIP 
destinations are 
perfect.  We also have several SPA-1001s and SPA-2000s that have been running 
perfect since day 
1.
 
 Also Sipura support is nonexistant.  Just our experience.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Dan
 Littlejohn
 Sent: Wednesday, June 15, 2005 9:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000?
 
 
 I have only had experience with the Sipura 3000 and I would agree with
 the voice volume problems.  I have given up on it working properly
 (adjusted gains, impedences, firmware, etc), the voice quality is just
 to low to actually use.  I actually purchased a second one thinking
 that the first might be defective.
 
 Would not recommend it because of the low sound volume problem. 
 Talking on the phone is actually the point of the device so who cares
 how configurable it is if you cannot hear anything.  I purchased a
 Digium TDM400P and have had very good luck with it.
 
 Dan
 
 On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote:
   Just want to tap the collective wisdom of this list as to experiences
   pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
  
  I've not played with the ht488, but I believe others have posted this
  device does not provide access to the pstn-fxo port. The spa3k does
  provide that access (if you want it).
  
   Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
   the top of the pick..Any comments and experiences esp. with Asterisk
   compatibility would be great, before I plonk in the bucks.
  
  The spa3k works fine with asterisk as many have posted. However, once
  in awhile it does act a little strange in two different ways:
   1. the spa3k will sometimes interpret some voices as tones which cause
   a little disturbance to any conversation going on. It is sort of like
   the old telephony talk off that existed years ago. Doesn't happen
   all that often and seems to be more sensitive to female voices based
   on my one-year of experience.
   2. sometimes it seems to operate in half-duplex mode, where if you try
   to talk at the same time as the other end is talking, the other end
   won't hear you.
  
  Neither one of those have been all that objectionable to me, but they
  happen and others have posted roughly the same issues. I've not heard
  of anyone that has found a way to minimize those two issues.
  
  The down side of the spa3k right now is that Cisco bought the company
  and there likely won't be much advancement of the code until after the
  ownership (and development efforts) are sorted out by both companies.
  (The same kind of product delays has been seen with their Linksys
  purchase, as well as when other companies are bought/sold.)
  
  Its fairly common knowledge that ex-Cisco folks started Sipura for the
  sole purpose of selling the company for a hugh profit. Their success
  in accomplishing that objective could only be measured in terms of
  producing Sipura products that had at least some acceptance of those
  products by end users. With those previous objectives accomplished,
  how will Cisco handle the Sipura products in the future? (It's any-
  one's guess at this point since Cisco also has at least some track
  record of mismanaging purchased companies for whatever reason.)
  
  From an internal Cisco 

RE: [Asterisk-Users] HT-488 vs. SPA-3000?

[Tarpo, Louie]

The only thing that snaps to mind is the interesting wiring in the building.  
The phone lines come into a 66 block, then are jumped across 3 more 66 blocks, 
then over to a 110 block, then down to another 110 block, then finally down 
into the Sipura3000s.  The 66 and 110 blocks feed the old analog system around 
the building.  The lines aren't connected to phones at the other ends, but they 
were left in place as a fallback position.  It's a manufacturing facility so I 
suspect there could be some interference in the building crossing into the 
phone lines.  I'm planning on wiring directly from the Qwest 66 block into the 
SPA3000s and disconnecting the rest of the phone system on my next visit.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Littlejohn
Sent: Wednesday, June 15, 2005 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000?


Perhaps there is something else going on the the Sipura 3000.  Its
voice quality and volume so poor/low that the device FXO port is not
usable.  However, same everthing and the TDM400P card works perfectly
with excellent voice quality and volume.  My experience, obviously
just one data point.


On 6/15/05, Tarpo, Louie [EMAIL PROTECTED] wrote:
 I'm curious what other standalone FXO adapters work with Asterisk.  At 
 everything from the default to the maximum in positive and negative values, 
 and combination of gain settings, we still get unacceptable distortion and 
 echo.  I've checked the phone lines, they work normally with a regular phone.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Rich
 Adamson
 Sent: Wednesday, June 15, 2005 1:05 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] HT-488 vs. SPA-3000?
 
 
 The majority of the audio level issues seem to be on the fxo port
 and setting the transmission levels (gain) to compensate for the
 cable loss to the central office. Eg, setting the pstn gain values
 to what should be appropriate causes echo, etc, not unlike the TDM
 card. (I have both in use.)
 
 In other words, the further the spa3000 (or TDM card) is from the
 central office, the more difficult it seems to be to set gain values
 that are acceptable. That's apparently why many people find its use
 is okay while others seem to think its objectionable.
 
 
  We have 6 SPA3000s.  The device is extremely configurable and works 
  inbound/outbound with
 Asterisk with the latest firmware update with little trouble.  However, we've 
 yet to resolve
 sound volume and quality issues.  The PSTN to SPA gain and SPA to PSTN gain 
 along with FXS Port
 Input Gain and Output Gain settings have had no positive effect.  The problem 
 is entirely with
 the analog line adapter.  VoIP calls from the analog phone to other VoIP 
 destinations are
 perfect.  We also have several SPA-1001s and SPA-2000s that have been running 
 perfect since day
 1.
 
  Also Sipura support is nonexistant.  Just our experience.
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Dan
  Littlejohn
  Sent: Wednesday, June 15, 2005 9:55 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000?
 
 
  I have only had experience with the Sipura 3000 and I would agree with
  the voice volume problems.  I have given up on it working properly
  (adjusted gains, impedences, firmware, etc), the voice quality is just
  to low to actually use.  I actually purchased a second one thinking
  that the first might be defective.
 
  Would not recommend it because of the low sound volume problem.
  Talking on the phone is actually the point of the device so who cares
  how configurable it is if you cannot hear anything.  I purchased a
  Digium TDM400P and have had very good luck with it.
 
  Dan
 
  On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote:
Just want to tap the collective wisdom of this list as to experiences
pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
  
   I've not played with the ht488, but I believe others have posted this
   device does not provide access to the pstn-fxo port. The spa3k does
   provide that access (if you want it).
  
Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
the top of the pick..Any comments and experiences esp. with Asterisk
compatibility would be great, before I plonk in the bucks.
  
   The spa3k works fine with asterisk as many have posted. However, once
   in awhile it does act a little strange in two different ways:
1. the spa3k will sometimes interpret some voices as tones which cause
a little disturbance to any conversation going on. It is sort of like
the old telephony talk off that existed years ago. Doesn't happen
all that often and seems to be more sensitive

RE: [Asterisk-Users] Includes include the includes?

[Tarpo, Louie]

Yes it does.  You want something like this...

[office]
include = default
include = local
include = international

[building1]
include = default
include = local

[default]
exten = 700,1,Dial(SIP/${EXTEN})
exten = 100,1,Dial(SIP/${EXTEN})
exten = 200,2,Dial(SIP/${EXTEN})
;and so on for your other extensions

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Mason
(Lists)
Sent: Wednesday, June 15, 2005 6:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Includes include the includes?


I am grouping my extensions by building like so:
1XX  is Building 1
2XX  is Building 2
7XX  is Office

[Office] extensions has the following includes
7xx
Include = Local
Include = International
Include = Building1
Include = Building2

[Building1] has 
1xx
Include = Office
Include = Building2
Include = Local

I done't want building1 to access international, but does it inherit that
include through including the office context?


Chris Mason

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RE: [Asterisk-Users] Best BootRom SIP Code for Poly600?

[Tarpo, Louie]

I'm using bootrom 2.6.1 with 1.5.2 for the same reason.  I would suggest the 
upgrade to 1.5.2 for some non trivial enhancements such as multiple line/call 
appearance.  Also the menu system is significantly improved.

Louie

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ariel
Batista
Sent: Friday, June 10, 2005 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best BootRom  SIP Code for Poly600?


Justin Ellison wrote:
 Hey all,

 Just getting started playing around with my Polycom 600.  According to
 the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP
 1.4.1.  Is that info still current, or is it safe to upgrade to 3.0.1
 and 1.5.2?

I am still running BootRom 2.6.1 with Firmware 1.5.2 works great. I don't 
want to upgrade the rom due to not being able to down grade.



 Justin 
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