[asterisk-users] MWI from ITSP

[Tom Vile]

How do I assign the MWI to a SIP phone on my asterisk server that is coming
from an ITSP?

I see the SIP message come across as having a message waiting but how does
one get that
to go to an extension on my box.

Thanks

Tom
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[asterisk-users] gotoiftime and blocking calls

[Tom Vile]

I am trying to use the Gotoiftime CMD to not allow calls to be placed
between the hours of 12am-5am, except if you know the PIN number to dial out
and if the call is for 911.

What is the best way to implement this solutions?

I have the gotoiftime like so:

exten = s,1,GotoIfTime(5:00-11:59|mon-fri|*|*?custom-blacklist,s,1)

and using Read for the PIN like so:

exten = s,3,Read(Secret,,3)
exten = s,4,NoOp(${Secret})
exten = s,5,Gotoif($[${Secret} = 123]?6:8)

but I guess I am stuck at allowing 911 calls to go through and what order to
place them in.

Thanks for the suggestions.


Tom
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Re: [asterisk-users] trixbox + agi

[Tom Vile]

yes, you can use Trixbox.On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote:
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do the php+agi integration, I'll do that, if not, will try to just try to install Asterisk from source.
Thanks!On 11/15/06, Tim Uckun 
[EMAIL PROTECTED] wrote:
If I were you I would go the AGI way. Use ruby, python, php, perl,java, c# or even erlang. Aything but the asterisk dialplan commands.There is no sense in putting yourself through that pain.___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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[asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error:Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format
But the file does exist and I see the entries for the key presses that I put in but it will not stream the file back to me using Cepstral.Asterisk 1.2.9CentOS 4.2Thanks,Tom Vile

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Re: [asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

There are no file extensions. It is just-rw-r--r-- 1 asterisk asterisk 32 Nov 15 12:52 swift_082da06a422be49e3a475925d9fc50e6-rw-r--r-- 1 asterisk asterisk 7 Nov 15 12:52 swift_6fc422233a40a75a1f028e11c3cd1140
-rw-r--r-- 1 asterisk asterisk 13 Nov 15 12:52 swift_80339585692b0188288da14748213dcc-rw-r--r-- 1 asterisk asterisk 11 Nov 15 12:54 swift_f87b365372c500c76e497087ac7e470a
On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:
Are you including the file extension?JayTom Vile wrote: I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error:
 Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format But the file does exist and I see the entries for the key presses that I
 put in but it will not stream the file back to me using Cepstral. Asterisk 1.2.9 CentOS 4.2 Thanks, Tom Vile 
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

It does error when I run it from the CLI with error:

sh: -p: command not found

I am assuming that it is referring to this line in phpagi.php

shell_exec({$this-config['cepstral']['swift']} -p
audio/channels=1,audio/sampling-rate=$frequency $voice -o $fname.wav -f
$fname.txt);

On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:


1) Try giving it an extension (say .gsm) and seeing if that works.  Make
sure you change both the file and your script.

2) Does the rest of the script work?  If you run './test.php', do you
get any errors?

Jay

Tom Vile wrote:
 There are no file extensions.  It is just

 -rw-r--r--   1 asterisk asterisk   32 Nov 15 12:52
 swift_082da06a422be49e3a475925d9fc50e6
 -rw-r--r--   1 asterisk asterisk7 Nov 15 12:52
 swift_6fc422233a40a75a1f028e11c3cd1140
 -rw-r--r--   1 asterisk asterisk   13 Nov 15 12:52
 swift_80339585692b0188288da14748213dcc
 -rw-r--r--   1 asterisk asterisk   11 Nov 15 12:54
 swift_f87b365372c500c76e497087ac7e470a


 On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:

 Are you including the file extension?

 Jay

 Tom Vile wrote:
  I am trying to get the example input.php working from PHPAGI but it
 will
  not
  playback the letters that I put in because of this error:
 
  Nov 15 14:25:22 WARNING[18678] file.c: File
  /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any
 format
 
  But the file does exist and I see the entries for the key presses
 that I
  put
  in but it will not stream the file back to me using Cepstral.
 
  Asterisk 1.2.9
  CentOS 4.2
 
  Thanks,
 
  Tom Vile
 
 
 


 
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?

[Tom Vile]

They brake easy.
Speaker phone is not very good.
Overall sound not good compared to a Snom, Polycom or Cisco phone.
Drop registrations with Asterisk randomly.
Power supplies die.  Had 4 out of 10 go bad within a year.
LCD backlight died on 2 that I deployed.

We only do the Snom 320 or 360's now and are just as easy to configure and
have alot of great options as well.

On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote:




We are doing a medium sized office in NYC with 80 phones. The customer
originally requested Polycom 601 phones. The COO also authorized us to
purchase 2 Grandstream GXP2000 phones for the mail room. We find these
phones much easier to configure and work with asterisk . They support BLF 
intercom right out of the box. They can also be centrally managed and
provisioned. They also sound great and work in a very intuitive way. We
don't have real life experience deploying this phone so I'm just going to
ask:



Is there a catch?  Why the huge price difference? These phones seem to do
everything a busy corporate office would need. Is there a big qualitative
difference between this phone and Polycom501/601?? Is there a major problem
with this phone not disclosed by the manufacturer or vendors. Some feedback
from people who have deployed them would be great.



Thanks In advance.



JR

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--
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Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?

[Tom Vile]

Yes, they support both and quite nicely.  The web interface for the phone is
very good and you can program alot of the buttons to do what you want (ie)
The record button can be setup to do 1 touch recording and such.

On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote:


 Thanks for the input.  I take it the snoms support both BLF  intercom?





==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238 http://172.17.9.10/clicktocall.php?number=19173327238
Office: 212-624-5943http://172.17.9.10/clicktocall.php?number=12126245943
Web: www.euscorp.com
==
  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Tom Vile
*Sent:* Wednesday, November 15, 2006 9:16 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?



They brake easy.
Speaker phone is not very good.
Overall sound not good compared to a Snom, Polycom or Cisco phone.
Drop registrations with Asterisk randomly.
Power supplies die.  Had 4 out of 10 go bad within a year.
LCD backlight died on 2 that I deployed.

We only do the Snom 320 or 360's now and are just as easy to configure and
have alot of great options as well.

On 11/15/06, *Jeronimo Romero* [EMAIL PROTECTED] wrote:



We are doing a medium sized office in NYC with 80 phones. The customer
originally requested Polycom 601 phones. The COO also authorized us to
purchase 2 Grandstream GXP2000 phones for the mail room. We find these
phones much easier to configure and work with asterisk . They support BLF 
intercom right out of the box. They can also be centrally managed and
provisioned. They also sound great and work in a very intuitive way. We
don't have real life experience deploying this phone so I'm just going to
ask:



Is there a catch?  Why the huge price difference? These phones seem to do
everything a busy corporate office would need. Is there a big qualitative
difference between this phone and Polycom501/601?? Is there a major problem
with this phone not disclosed by the manufacturer or vendors. Some feedback
from people who have deployed them would be great.



Thanks In advance.



JR


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--
Tom Vile

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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[asterisk-users] Anyone using the directory.agi app in AGI perl

[Tom Vile]

Just wondering if anyone is using the directory.agi app in AGI perl, I can't
seem to get the searching function working.  It lists my contacts but when I
type in digits it say no match found.

Thanks

--
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Re: [asterisk-users] trixbox + agi

[Tom Vile]

On 11/14/06, Tim Uckun [EMAIL PROTECTED] wrote:
On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote: I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row
 at a time back to the caller.First, is this beyond Asterisk?Second, can I do this if I use the Trixbox implementation?Third, any good tutorials on doing just this?There are numerous AGI toolkits in different languages. I have just
started fooling around with RAGI which is integrated with ruby onrails.From my experiments so far it seems to work OK.___--Bandwidth and Colocation provided by 
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I do this currently but dont know how to allow additional rows to be
read back. It only reads the first one then stops. Any insight on how
to have it loop to get addtional rows would be great. Also I was wondering how to get DTMF Digits to translate in to possible matches for LastName to query my database like the Directory CMD does.

[custom-lookup]
exten = s,1,Cepstral(Please enter your eye dee Number, followed by the pound key now.)
exten = s,2,Read(ID)
exten = s,3,MYSQL(Connect connid localhost contacts  contacts)
exten = s,4,MYSQL(Query resultid ${connid} SELECT\
FirstName\,LastName\,HomePhone\ FROM\ contacts\ WHERE\
ContactID=\'${ID}\')
exten = s,5,MYSQL(Fetch foundRow ${resultid} var1 var2 var3) ; fetch row
exten = s,6,GotoIf($[${foundRow} = 1]?7:9) ;
exten = s,7,Cepstral(The Phone number for ${var1} ${var2} is, ${var3}.)
exten = s,8,Goto(s,10)
exten = s,9,Cepstral(No match found. Goodbye) ; End loop if
exten = s,10,MYSQL(Clear ${resultid})
exten = s,11,MYSQL(Disconnect ${connid})

If you need more help with the above let me know.Tom Vile
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Re: FW: [asterisk-users] Desktop integration

[Tom Vile]

asteriskextras.com has a FREE click to call script that is very popular.

On 11/13/06, mitcheloc [EMAIL PROTECTED] wrote:

Snap will do this for you. (Check my signature)

On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote:

  Hi Dean,

  I will check that site - thanks for the hint.
  The biggest problem I see with authentication and I do not think mexuar
 could help me here (and I am definitely going to pay $2000 for it :-)
  But it is another story...

  Thank you!
  Ondrej

  Dean Collins wrote:



 Ondrej,

 You could do it using Mexuar Corraleta but this is a commercial application
 for Asterisk (US$2,000)

 http://www.mexuar.com/products_sdk.shtml

 http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf



 However it has a whole heap more functionality than what you are looking
 for.



 If you just want to do 2 legged outbound calls check out 'call files' on
 www.voip-info.org








 Cheers,



 Dean




  


 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf
 Of Ondrej Valousek
  Sent: Monday, 13 November 2006 6:29 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Desktop integration



 Hi all,

  I am interested in integrating my telephone system (I am using hardphones
 and Asterisk) with my desktop - something like this:

  1. someone sends me his/her phone number via email/icq
  2. I cut/paste the number in some application/web page (?)
  3. my phone starts ringing and when I pick it up I will get connected with
 the remote party.

  Now I know I have read some discussion about this possibility but I can not
 recall where.
  Many thanks for any point.

  Ondrej 

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--

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com
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[asterisk-users] Question about MySQL Fetch foundRow from the dial plan

[Tom Vile]

I have a query that query's my database based on the read input for an ID number.exten = s,4,MYSQL(Query resultid ${connid} SELECT\ `FirstName`\ `HomePhone`\ FROM\ `contacts`\ WHERE\ `ContactID`= \'${ID}\')
exten = s,5,MYSQL(Fetch foundRow ${resultid} var1 var2) ; fetch rowProblem is is that when I do the following:exten = s,7,Cepstral(The Phone number for ${var1} is ${var2})is reads back
Executing Cepstral(SIP/2092-ef65, The Phone number for John Doe is ) in new stackThe var2 is not displaying. If I take out FirstName from the query it works fine.Any suggestions?
-- Tom Vile6
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[asterisk-users] Application Directory question

[Tom Vile]

I was wondering how the Directory CMD can read the input of numbers from a phone and translate that to search in voicemail.conf. Essentially I want to be able to look up contacts with MySQL and have the user input 3 digits corresponding to the contacts last name and have it search for it in the database.
I already have the connection into mysql working but just wondering how I would go about getting the numeric input to be translated to TEXT.Thanks for the pointers.Tom Vile
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Re: [asterisk-users] (no subject)

[Tom Vile]

Add a subject next time.

Are you behind a firewall where the Asterisk server is located?  Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?

On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:


i am sure this came up before
but all my searches are not resulting in anything usefull

trying to setup a grandstream phone
to connect to an asterisk server

now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port

in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036,   20001, 2727}

---
on the server side

same thing
plus
voip_users =  ip from where i am connecting 

--
can't seem to find anything else that should be opened on either side
to allow connection


--
i guess, help ?
--


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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [asterisk-users] Re: Port Range

[Tom Vile]

That probably because you are using Webmin. Just change the port Webmin listens on instead, I use 9000.On 11/6/06, Zeeshan Zakaria 
[EMAIL PROTECTED] wrote:I'll keep that in mind for future. I read about using 10001 as start port on Nerd Vittles website.


Is there some good material online to read more about RTP, SIP, RTCP and UTP?

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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] Re: Port Range

[Tom Vile]

Yes, it can. I put all of my servers to 9000 though.On 11/7/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Webmin uses UDP?Audio is generally RTP over UDP.Tom Vile wrote: That probably because you are using Webmin.Just change the port Webmin
 listens on instead, I use 9000. On 11/6/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I'll keep that in mind for future. I read about using 10001 as start port
 on Nerd Vittles website. Is there some good material online to read more about RTP, SIP, RTCP and UTP?___--Bandwidth and Colocation provided by 
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] best gui

[Tom Vile]

He is not talking about Trixbox but FreePBX and his assumption is correct. Just load Asterisk and then FreePBX later.On 11/4/06, Zeeshan Zakaria 
[EMAIL PROTECTED] wrote:No, you can't add this to asterisk install. Trixbox formats the drive and installs everything from scratch. But it installs all the necessary software packages which one may need for a perfectly working phone system, and configures them. This saves a lot of time and headache and you can move on from there. Now Trixbox is mature enough to rely upon.


Once you are familiar with it, then you'll need to do your own customizations as well, for which you definitely need know how of Asterisk and Linux. But for the beginning, Trixbox is good.

During installation, trixbox downloads and updates asterisk, zaptel and many other software packages as well to the latest versions, or versions which are compatible with each other.

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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] FXO lines taking several rings to answer, always two

[Tom Vile]

If its always 2, then its waiting for CallerID or Fax Detection?On 11/4/06, Jordan Novak [EMAIL PROTECTED]
 wrote:They are in Kewl start now but I have tried groundstart and 
loopstart. Waht could i be missing that would cause this.I start with a 
Exten= s,1,answer. I am using three FXS modules on a 
tp400.
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] Snom or Cisco Phones?

[Tom Vile]

I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc [EMAIL PROTECTED]
 wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons
can be retooled for any environment, making custom installs very easy.On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote: On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
  Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia  cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new
 snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360)
 * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can
 park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos.
 * On ciscos, I find the upgrade path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by 
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--Mitchel ConstantinSnap - A desktop user interface for Asteriskwww.snapanumber.com___
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Tom Vile]

I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99.2 Asterisk boxes in different locations? Sure, you can do that and its quite easily.
On 11/1/06, Ken Williams [EMAIL PROTECTED] wrote:
Thanks everyone for the input.After pricing everything we need out,it's not worth trying to get our old system to work, so I've pitchedditching everything and starting over.I'm very excited and hopingthey'll go for it.
Regardless, I'm going to throw a box together for my house, we have nohome phone (just cell phones) so this'll be a great way of testing.All that being said, any comments on the Grandstorm phones?I've
ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERYinexpensive for a business solution.I see it has room for 4 lines with7 programmable buttons.I assume I can put a few more lines on the
programmable buttons (we have 6 lines at our main location).One last newbie question, I assume if I have an Asterisk PBX at 2locations in different states, I'll be able to transfer a call thatcomes into location1 to a user at location2.
Thanks again for the quick responses  help.-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of AndrewLathamSent: Wednesday, November 01, 2006 5:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Re: Newbie Questions
KenIf these are older comdials then they are just analog phones with extrasignaling.The extra signaling could be on the main twisted pair(likely) or on the next twisted pair as data (9600 baud modem) like some
of the nortels do.Always remember that it would cost the companies aton to make every system totally closedThat being said, the entry price for IP phones or ADSI phones can bemuch lower than you think.Find a good consultant in your area, get an
ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with.You can order the Aastra phones from your local electrical supplycompany (the place with a long counter and lots of electricians drinking
coffee ordering their parts.).AndrewOn 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could
 have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a
 VOIP connection.That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new
 hardware for everything, thought I'd see if a cheaters option wasavailable. Thanks for any help. ___ --Bandwidth and Colocation provided by 
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http://lists.digium.com/mailman/listinfo/asterisk-users-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] -
[EMAIL PROTECTED] If any of the above are down we have bigger problemsthan my email!Hind sight is most always 20/20 or better.---___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Tom Vile]

it told you:Permission DeniedCheck the permissions on that directory.On 10/31/06, Zeeshan Zakaria 
[EMAIL PROTECTED] wrote:Anybody knows why ARI gives this error message when I enter extension number and password.


Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525


It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX.
-- Zeeshan A Zakaria 

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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Tom Vile]

Did you install PHP?On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote:













Hi All,

 I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface. 

After installing properly when opening in the webpage it is
not parsing the index.php for the AMP. My Database is MySQL.and web server is
Apache 2.2.



Please let me know is this configuration problem or this is
the problem with Apache (Apache 2.2) .





Thanks and Regards

Alok Mohapatra











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Re: [asterisk-users] Compatability

[Tom Vile]

http://www.digium.com/en/docs/misc/compatibility_notes.phpServer Compatibility The following list of servers are known
to be partially incompatible with Digium hardware. We do not recommend
using the following computers to set up an Asterisk server: Dell PowerEdge 1600Dell PowerEdge SC 420Dell PowerEdge SC 1420Dell PowerEdge 650Dell PowerEdge 700Dell PowerEdge 750
Dell PowerEdge 2650IBM eSeries 336IBM xSeries 345IBM xSeries 360/365
			Motherboard Chipset Compatibility
			The following list of motherboard chipsets are known to be partially incompatible with Digium hardware:
			Intel 915 (all variations)Intel E7221Intel E7525
			Digium Hardware Motherboard Compatibility
Some server motherboards utilize an onboard Intel e1000 Ethernet
controller that can interfere with the operation of Digium's cards. The
recommended action for this server is to disable the onboard Ethernet
controller and use a PCI-based solution. Also, the MS-7032 (K8T
Neo-V/K8M Neo-V) motherboard is incompatible with the TE4XXP using the
firmware ending in 164. The problem is that the card will randomly
receive interrupts.On 10/31/06, Joel Hill [EMAIL PROTECTED] wrote:
Hi All,I have a new client who has an existingAsteriskPABX and is lookingfor us to install a TE110P for him, However he has a Dell SC420 and Ihave never used one before.I have had no problems with any other Dell servers which we use almost
exclusively.Has anyone had any good/bad experiences with the SC420 in relation withDigium cards?Thanks for your help.JoelAsterisk ITwww.asteriskit.com.au
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Re: [asterisk-users] Mac OS X Desktop / Asterisk integration?

[Tom Vile]

I used some of the ideas found here:http://www.voip-info.org/wiki/view/Asterisk+manager+ExamplesOn 10/30/06, 
Steve Davies [EMAIL PROTECTED] wrote:
Hi,We are successfuly using TAPI with Asterisk in order to provide ageneric and fairly well supported interface from Windows desktops toAsterisk - This allows caller-id popping and click-to-dial from TAPI
aware environments.Is there an equivalent telephony interface available for Mac OS X, andif so, is there an asterisk-manager plugin for it?Thanks for any feedback - All of the searching I have done has ended
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[asterisk-users] Forwarding recorded calls to Voicemail

[Tom Vile]

I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail?ThanksTom Vile
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Re: [asterisk-users] CID Issues

[Tom Vile]

On 10/23/06, mail-lists [EMAIL PROTECTED] wrote:
Hello,I've posted this at the trixbox and freepbx forums and haven't been ableto get an answer. I thought perhaps the guru's here might be able tohelp me out :)I'm having some issues with setting caller IDs. There are 2 problems
that I would like to solve.1. I have a DID pointing to a ring group. The only 'extension' in thatring group is an external number (cell phone). So essentially the DIDfwds to a cell phone. The problem is that the CID that shows up on the
Cell phone is the number that's set on the outgoing trunk the theCALLERS #. Is there a simple way to override this? Or better yet, isthere a prefered method for forwarding calls out with freePBX?
Are you allowed to set your own CallerID on outbound calls from your provider?
2. I have sevaral trixbox installs connected through DUNDI. The DUNDIworks very well.. I can call local extensions from every PBX. The PBX'sare connected via an IAX trunk. In freePBX I've created a custom trunk
that accepts a 4 digit extension and puts the call into a 'trydundi'context. The problem I'm having is that whenever someone calls from anextension at one location to an extension at another location theCallerID that shows up at the other location is the one set either in #1
The custom trunk, or #2 in the 'Outbound CID' field in the users screen.What I WANT this to be set to is the Name of the extension ie. just likelocal calls are. Is there a way to do this painlessly. Is it possible to
hook dundi into a different context so that it would think all calls arelocal.. I'm kinda guessing here.Sorry about the length of these descriptions and thanks for any advice!___
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Re: [asterisk-users] CID Issues

[Tom Vile]

The o option is mentioned over at FreePBX and how to restore this setting.
On 10/23/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
mail-lists wrote: Hello, I've posted this at the trixbox and freepbx forums and haven't been able
 to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve.
 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the
 Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX?
I cannot help you with Trixbox, only Asterisk.Are you using PRI, Analog, or VoIP for your outgoing call?If PRI or VoIP AND the carrier permits it, you can manually set theCaller*ID before the Dial() line using SetCIDNum.
You can also do a show application dial in the Asterisk CLI, payspecial attention to the o option to Dial()___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] Access Denied on a Windows share

[Tom Vile]

Is it an NTFS Share?On 10/19/06, Paul Gaffney [EMAIL PROTECTED] wrote:


















Message: 12



Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)



From: sdgesa gaeharth
[EMAIL PROTECTED]



Subject: Re:
[asterisk-users] Extremely choppy sound on some of




ourPOTSnetwork calls; goes away with mute



To: asterisk-users@lists.digium.com




Message-ID:
[EMAIL PROTECTED]



Content-Type: text/plain;
charset=iso-8859-1







None of these steps have
made a difference. Any other suggestions? Here is

my original post:









Can anyone help me to figure
out why I can not write to a public share? I

was able to join the domain
without a problem. I can access the share from

an xp box. I
just can not write: Access denied.



 



 thanks








If the person or process that is trying to write to the
share is a member of a group that is denied access then the write will fail
with "access denied". 



Check the "effective permissions" for a user on
the file and look at the server security logs which may give you some
additional information.



Paul Gaffney

LANStatus, LLC













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Re: [asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)

[Tom Vile]

not having that issue with ChanSpy here but I loaded up a 1.2.12.1 box last night with a TE110P and Asterisk Crashes after receviing a call and I was using the latest zap drivers. I put in the Sangoma card and no problem. Must have been some motherboard compatibility.
On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mauro Zanin wrote:Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in 
voicemail.conf, so it didn't work. Installed again 1.2.0 and voilà the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: Chi lascia la via vecchia per la nuova,sa quel che perde
 e non sa quel che trova litterally: Who leaves the old way, does know what he loses and doesn't know what he finds. Ciao MauroI've not had this problem, but I can say that with 
1.2.12.1 if I usechanspy, when the spying handset hangs up asterisk segfaults (kickingall connected calls off).Finding that out was embrassing. (was on production server).
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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?

[Tom Vile]

I 4th it.On 10/18/06, Matthew Thompson [EMAIL PROTECTED] wrote:
On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful.
  I've never had any problems with their products that wasn't my own fault.
 Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you)
[EMAIL PROTECTED] :o) 
--Matthew Thompson[EMAIL PROTECTED]
 
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Re: [asterisk-users] how to config chanspy

[Tom Vile]

seriously go look at voip-info.org for the answer, thats where we get most of our info from, or perhaps type show application chanspy from the asterisk CLI.Are we that lazy that we cant use google to search. Ridiculous.
On 10/18/06, Sergio R. D'Ippolito [EMAIL PROTECTED] wrote:















How can I do to select
the channel to spy ?

thanks











De: 
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] En nombre de 
Ralph Liebessohn
Enviado el: Miércoles, 18 de
Octubre de 2006 09:29 a.m.
Para: Asterisk
 Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] how
to config chanspy





On 10/17/06, Thirumal Saminathan
[EMAIL PROTECTED] wrote:








hi all,





please any one help me ,how to configure chanspy application .





and also send me if u have any sample configure file.

















-thiru









Hi,

It could be very simple, like:

exten = 123,1,ChanSpy()
; Spy all channels

or more accuracy:

exten =124,1,ChanSpy(SIP)
; Spy all sip channels 

if I can help you more, let me know!

-- 
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn 







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Re: [asterisk-users] Is 1.2.12.1 production ready

[Tom Vile]

fine for me here since it came out. We are running 15 extension all day long.On 10/16/06, shadowym [EMAIL PROTECTED]
 wrote:I am getting ready to image a production system.Right now I am planning on
using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3.I will be using aSangoma A200D card.I read of some people having problems with Asterisk 1.2.12.1
 crashing.Isthis across the board or is there anyone out there with no problems.If youhave 24/7 uptime and no nightly reboot crons I would definitely appreciatehearingabout it.Cheers___
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Re: [asterisk-users] Polycom HDVoice

[Tom Vile]

http://www.polycom.com/products_services/1,1443,pw-17068,00.htmlOn 10/13/06, 
Jessee J Holmes [EMAIL PROTECTED] wrote:
I've played with one, but they are not available yet from Polycom (as in shipping to distributors). They are only demoing the phones at the moment. I'm sure if anyone has gone to any VoIPtrade-show (like VON) they would have seen and used one of these, but under Polycom environments.
It uses the G.722 codec, this is HDVoice according to Polycom. It's supposed to sound better. Under Polycom demo conditions, it does sound better. Polycom has samples of the sound of the phones on their website I believe so you can hear for yourself, but I'm stillskeptical of what it really will sound like until we all get the phones in our hands and installed on our messed up Asterisk environments (joke). :)
Hope that helps some. 

Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/
 On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote:Has anyone used the Polycom HDvoice phone yet?
 I am curious if ituses a different codec. Does it actually sound any better?___
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Re: [asterisk-users] How big is *your* dialplan??

[Tom Vile]

315 extensions (1198 priorities) in 92 contexts.On 10/10/06, Jeremy McNamara [EMAIL PROTECTED] wrote:
One of the smaller systems:-= 9924 extensions (29772 priorities) in 6 contexts. =-Jeremy McNamara___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] How to make this easier

[Tom Vile]

Thanks James, I was close.On 10/8/06, James Jones [EMAIL PROTECTED] wrote:
exten = _*1XX,1,Set(CALLERID(all) = Nursery ${EXTEN:1})exten = _*1XX,2,Dial(SIP/400)Tom Vile wrote: I have a need for a dialplan that call for the ability for people to dial *1XX and it send a call
 to extension 400 with the calleridname of Nursery and the calleridnum of the *1XX number that was put in minus the *.Now I know how to do it individually but I now there must be an easier
 way to simply the code. Any help would be appreciated. Tom Vile___
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[asterisk-users] How to make this easier

[Tom Vile]

I have a need for a dialplan that call for the ability for people to dial *1XX and it send a callto extension 400 with the calleridname of Nursery and the calleridnum of the *1XX number thatwas put in minus the *. Now I know how to do it individually but I now there must be an easier
way to simply the code.Any help would be appreciated.Tom Vile
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Re: [asterisk-users] How do I reset a password?

[Tom Vile]

login as root and type help-aah and you will see a list of commands to change the admin password.On 9/30/06, Jim Lynch 
[EMAIL PROTECTED] wrote:I'm looking for the username/password to access the web gui for freepbx
admin rather than the voicemail passwords.I need to reconfigure theextentions/ring groups.Thanks,Jim.Doug Lytle wrote: Jim Lynch wrote: I've forgotten the user/pw for my freepbx adnim.I'm using
 [EMAIL PROTECTED]Is there a way to discover them or reset them.I have root access to the system.I did a google search but that didn't help. On a normal installation the passwords are located in the
 voicemail.conf file located in /etc/asterisk/voicemail.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] trixbox t38 pass through

[Tom Vile]

On 9/25/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
FYI, this is an asterisk mailing list, NOT trixbox. Most people on here don't care when trixbox is going to do something. Try their list.
On 9/25/06, Christopher Corn 
[EMAIL PROTECTED] wrote:
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for now i will have to setup my fax machine to connect directly to my t38 provider. anyone know when trixbox may have this update? 
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What do you think Trixbox is based on? ASTERISK.
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Re: [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right

[Tom Vile]

They only do numeric callerid.On 9/11/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
I have tested Grandstream Budgetone 102 and Grandstream Budgetone200 and with both, if they are called from a caller that is analphanumeric user, their display shows a unintelligible name impossible
to figure out who is calling!! If the caller is a numeric one, in bothphones their display shows correctly the caller's contact.I've updated their firmwares to the latest ones and that problem persists...
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Re: [asterisk-users] does anyone offer truly unlimited voip in the US

[Tom Vile]

Teliax is not unlimited but has a cap of 2500 minutes per month.***
Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable).On 8/30/06, Crazy Boy 
[EMAIL PROTECTED] wrote:Hi,  Taliax has unlimited calling plan per month. You can see 
WWW.TELIAX.COM  Regards, Chandra.Steven M. Sawczyn 
[EMAIL PROTECTED] wrote:
 Greetings, I finally  got my Asterisk server up and running and now am in the process of looking for a  provider to use as a SIP trunk. Unfortunately, I'm realizing that  unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for  example, translates to a mere 2500 minutes/month. In researching other SIP  providers, I'm finding that their terms of service define unlimited as  something similar. Does anyone know of a provider in the US that turly 
 offers unlimited calling, or segnifigantly more than 2500  minutes/month?  Thanks for any  suggestions,
  Steve 
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		Do you Yahoo!? Everyone is raving about the 
 all-new Yahoo! Mail.
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Re: [asterisk-users] does anyone offer truly unlimited voip in the US

[Tom Vile]

$24 per monthOn 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:





What 
cost do you pay per month for the 2500 minutes?

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]]On Behalf Of Tom 
  VileSent: 30 August 2006 13:54To: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does 
  anyone offer truly unlimited voip in the USTeliax is not 
  unlimited but has a cap of 2500 minutes per month.*** Softcap of 
  2500 Minutes (including 1000 minutes of toll-free inbound, if 
  applicable).
  On 8/30/06, Crazy 
  Boy  
  [EMAIL PROTECTED] wrote:
  
Hi,Taliax has unlimited calling plan per month. 
You can see WWW.TELIAX.COMRegards,Chandra.
Steven M. Sawczyn  [EMAIL PROTECTED] wrote:



Greetings, I finally got my Asterisk 
server up and running and now am in the process of looking for a provider to 
use as a SIP trunk. Unfortunately, I'm realizing that unlimited really 
is in fact limited -- Galaxy Voice's unlimited plan, for example, translates 
to a mere 2500 minutes/month. In researching other SIP providers, I'm 
finding that their terms of service define unlimited as something 
similar. Does anyone know of a provider in the US that turly offers 
unlimited calling, or segnifigantly more than 2500 
minutes/month?


Thanks for any suggestions, 


Steve

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  518-631-2855 x205Fax: 518-631-2856 


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Re: [asterisk-users] Nobody is responding. Why? (Implement music on transfer)

[Tom Vile]

Use the m option at the end of the dial string. Google told me so.On 8/26/06, Crazy Boy [EMAIL PROTECTED]
 wrote:  Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:
1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra.
 
		 All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.
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Re: [asterisk-users] Can budgetone 101 display name part of cid?

[Tom Vile]

The Budgetone only supports a 12-digit caller ID LCDOn 8/15/06, Guus Houtzager [EMAIL PROTECTED] wrote:
On Tuesday 15 August 2006 17:53, Julian Lyndon-Smith wrote: Try without the 
 Set(CALLERID(all)=some text 123-123-1234Tried that, no effect, still shows only the number part, without the '-'sthough.I find it weird I should have to Set it explicitly, why does a softphone like
sjphone show what's in the callerid field of the phone in sip.conf (as Ithink it should be) and the budgetone does not? Implementation difference insip protocol? JulianRegards,Guus
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Re: [asterisk-users] Snakes On A Plane using Asterisk?

[Tom Vile]

Did it about 10x to friends. Pretty funny.On 8/10/06, Hugh L. Johnson [EMAIL PROTECTED] wrote:
http://snakesonaplane.varitalk.com
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Re: [asterisk-users] Phone Newbie Questions

[Tom Vile]

Yes it supports PRIOn 8/9/06, Colin MacMillan [EMAIL PROTECTED] wrote:
Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link - 

http://www.oreilly.com/catalog/asterisk/the pdf version can be found here but having one to read is much better.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
With a background in Linux, once you learn a bit more setting up asterisk will be easy. And yes Asterisk can handle a 100+ users with relatively standard hardware.Colin

On 8/9/06, Brian Becker [EMAIL PROTECTED] wrote:

First let me just say that I am a total newbie when it comes to phonesbut I have several years of Linux and it experience.I have beentasked with offering a competing solution to our current phoneproviders based on asterisk...
To show my complete ignorance I am going to try and describe outsetup, or as much as I know about it.We have a PRI line with asingle number that rolls over to several lines.We also have ahandful of analogue fax lines (are these part of the PRI...I dunno?).
There are also a few 800 numbers...(not sure if that fact reallymatters to our phone system or not).From my bit of research I ampretty sure asterisk does support PRI but please correct me if I amwrong 'cause none of this matters if it doesn't...
First is asterisk really capable of supporting a 100+ user base of phones?I understand (or at least think I do) that I need one of the DigiumDigital TDM Cards and that is where the PRI connection gets pluged
into...how then do I connect the asterisk system to the phone networkthat exists in the office? (Told you I was clueless)Any guidance would be appreciated or even links to an introduction onphone systems would be great (pictures would help too ;) ).
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Re: [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?

[Tom Vile]

http://www.thetechguide.com/howto/asterisk/chanbluetooth.htmlOn 8/8/06, 
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Has anyone used a bluetooth phone (eg Cingular 8125 aka HTC Wizard) as anFXS/FXO with asterisk?I'd like to be able to route incoming cellphone calls into asterisk, aswell as make outgoing calls from asterisk to the GSM network using the
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Re: [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?

[Tom Vile]

Does your cingular 8125 have bluetooth? If so it should work. Mine works with Motorola and LG phones fine.On 8/8/06, 
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Have you used it with a cingular 8125?-DanOn Tue, 8 Aug 2006, Tom Vile wrote: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
 On 8/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Has anyone used a bluetooth phone (eg Cingular 8125 aka HTC Wizard) as an
 FXS/FXO with asterisk? I'd like to be able to route incoming cellphone calls into asterisk, as well as make outgoing calls from asterisk to the GSM network using the phone and bluetooth.
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Re: [asterisk-users] VoipNow 1.2.0 Beta

[Tom Vile]

Did you look on the site?http://www.4psa.com/products/voipnow/demo.phpOn 7/29/06, Dinesh Nair
 [EMAIL PROTECTED] wrote:
On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think.As
 powerful as the config files, and command line interface is, there isis there anywhere we can take a look at screenshots without having todownload the entire package ?--Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/+==oOO--(_)--OOo==+
| for a in past present future; do|| for b in clients employers associates relatives neighbours pets; do || echo The opinions here in no way reflect the opinions of my $a $b.|
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Re: [asterisk-users] Grand stream 2000 will not dial *xx

[Tom Vile]

Turn off the call features in the phone, by default the *70 codes are enable in the phone so that the phone can do call waiting and such. If you want asterisk to do this you need to disable the feature codes in the phone.
On 7/28/06, Chris Bagnall [EMAIL PROTECTED] wrote:
 I just bought a grand stream 2000.It appears that it will not dial any number with a leading *(*70,*71) So I can not dial any of my Apps in *What firmware version are you running? We have plenty of GXP2000s in
clients' premises with plenty of numbers beginning * without any problems.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons
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Re: [asterisk-users] Grand stream 2000 will not dial *xx

[Tom Vile]

does it reach the asterisk console? and have you turned off the dial features in the phone?On 7/28/06, Cavanna, Richard 
[EMAIL PROTECTED] wrote:Here is the software version:Program-- 
1.1.0.16Bootloader-- 1.1.0.1When I pick up the line and dial *70 it just disappears and never dials.If I enable early dial it does dial *70 but then it breaks my outbound
routes.ThanksRich I just bought a grand stream 2000.It appears that it will not dial any number with a leading *(*70,*71) So I can not dial any of my Apps in *What firmware version are you running? We have plenty of GXP2000s in
clients' premises with plenty of numbers beginning * without anyproblems.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons
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Re: [asterisk-users] Source Directory of ASterisk

[Tom Vile]

/usr/src/asteriskOn 7/28/06, Wasif [EMAIL PROTECTED] wrote:
Hi,I am using TriBox 1.1.1/Asterisk. I want to know where I can find sourcedirectory of Asterisk in system so I can install Asterisk audio conversionmodule (
http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw promptsinto g729 prompts. It requires to point Asterisk source Include directory.Thanks___
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Re: [asterisk-users] How to connect 2 AAH

[Tom Vile]

you can trunk the two boxes together with IAX. Check out trixbox.org and search, its been covered a few times.On 7/21/06, 
Gidean Chan [EMAIL PROTECTED] wrote:







Hi!
Does anyone know how to connect 2 AAH 
IPPBXs so that one extension in A IPPBX can use the PSTN trunk in B IPPBX for 
dial out?

Thanks very much
Gidean

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Re: [asterisk-users] A very lost newbie.

[Tom Vile]

http://www.voip-info.orghttp://asteriskguru.org/tutorials/On 7/20/06, 
David R. [EMAIL PROTECTED] wrote:
Here's my situation. I'm a programmer at the ISP I work for and my supervisors have seen fit for me to become an administrator for our new ventures into the VoIP world. All of us are relatively new at this jazz, though naturally my higher ups have more experience than I. We're toying with VoIP a lot in the office, and I personally feel like I'm fairly far behind them in knowledge of it all.
I've done some studying the past day or two, and I grasp the basic principles of VoIP, Asterisk, and SER/OpenSER, and about 5% of the hardware involved in all of it (programmer; software guy). I'd like to catch up to them as quickly as possible so I can be of most use. Here's what I have available to me, and what I've done so far:
AVAILABLE:- A box with OpenBSD (my preferred OS for most production servers)- Asterisk installed on the box.- One IP PhoneDONE:- Installed Asterisk.- Written this e-mail. :)I'm wondering where to begin. I have Asterisk installed but don't know how to go about getting my IP phone to work with it. To start, I'd like to make calls only internally on my home network (not even screwing with my company's network yet; they're further along into configs than I want to get into just yet) to a soft phone that Ubuntu Dapper Drake (my OS at home) comes with.
My question is this:Where can I find good starter documentation(s) for my purposes?Thank you,David

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Re: [asterisk-users] Dlink DVG 1120S/Asterisk VoIP to PSTN

[Tom Vile]

I believe when you use the SIP firmware that the PSTN side does not
work like you are wanting.  There was a thread over on DSLREPORTS
about this last week.

On 7/17/06, David Freeman [EMAIL PROTECTED] wrote:

Okay, I've looked for some days and only ever find other people asking this
question and no responses (I did see a few e-mail me answers, but those
haven't panned out for me).

Is it possible to use a D-link DVG 1120M (flashed to SIP firmware version v
0.0-S08) to make PSTN calls from Asterisk?

I can use the device with two phones as SIP extensions, and I have it
configured to allow the phones attached to the ATA to dial # and get a
dialtone on the PSTN.

It seems I am close, but something is not clicking with me.

Any help is appreciated.

Thanks,
Sugar Dave Freeman

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Re: [asterisk-users] How do you harden an Asterisk install?

[Tom Vile]

For the NIC setup you can bond 2 cards together for redundency.  Take
a look here for some more info on bonding.

http://www.redhat.com/docs/manuals/enterprise/RHEL-4-Manual/ref-guide/s1-networkscripts-interfaces.html#S2-NETWORKSCRIPTS-INTERFACES-CHAN

On 7/13/06, shadowym [EMAIL PROTECTED] wrote:


I remember reading a small write up somewhere.  I think it was on the
Asterisk Wiki.  I can't find it anymore.  It's probably a bit dated by now
but some of it would still be relevant.

Can anyone recommend a good guide or even some of their own suggestions.

For clarity, what I mean by hardening is to make an Asterisk Server or
network appliance or embedded server or whatever you want to call it, as
fail safe, stable, and reliable as possible.  Just like an expensive
traditional PBX.  This is for a small business application of 50 extensions
or less.  It can't be too crazy like redundant servers or anything like
that.  I am looking for ideas like RAID 1, redundant power supply, cron job
to reboot every night (yuck!), disable caching(?), Astlinux on embedded with
CF, yada yada!

Anyway to set up automatic failover to a second Network Card with same IP if
primary network card fails?  That is one point of failure I haven't found a
way around yet.  Failure of the managed switch is another one I get a bit
paranoid about.  Switches generally don't fail but I'd like to have some
sort of fail safe plan.
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Re: [asterisk-users] How do you harden an Asterisk install?

[Tom Vile]

For the NIC setup you can bond 2 cards together for redundency.  Take
a look here for some more info on bonding.

http://www.redhat.com/docs/manuals/enterprise/RHEL-4-Manual/ref-guide/s1-networkscripts-interfaces.html#S2-NETWORKSCRIPTS-INTERFACES-CHAN

On 7/13/06, shadowym [EMAIL PROTECTED] wrote:

Thanks for the suggestions but I specifically asked for options OTHER than a
second server.  Your suggestions about disabling un-needed services are good
though.  I already do that.  I am hoping someone has some suggestions that
are not as obvious that I have perhaps not thought of.

 -Original Message-
 From: Warren (mailing lists) [mailto:[EMAIL PROTECTED]
 Sent: Thursday, July 13, 2006 12:36 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] How do you harden an Asterisk install?

 shadowym wrote:
 
  I remember reading a small write up somewhere.  I think it
 was on the
  Asterisk Wiki.  I can't find it anymore.  It's probably a
 bit dated by
  now but some of it would still be relevant.
 
  Can anyone recommend a good guide or even some of their own
 suggestions.
 
  For clarity, what I mean by hardening is to make an
 Asterisk Server or
  network appliance or embedded server or whatever you want
 to call it,
  as fail safe, stable, and reliable as possible.  Just like an
  expensive traditional PBX.  This is for a small business
 application
  of 50 extensions or less.  It can't be too crazy like redundant
  servers or anything like that.  I am looking for ideas like RAID 1,
  redundant power supply, cron job to reboot every night (yuck!),
  disable caching(?), Astlinux on embedded with CF, yada yada!
 
  Anyway to set up automatic failover to a second Network
 Card with same
  IP if primary network card fails?  That is one point of failure I
  haven't found a way around yet.  Failure of the managed switch is
  another one I get a bit paranoid about.  Switches generally
 don't fail
  but I'd like to have some sort of fail safe plan.
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 You are talking about 2 things:
 (1) How to harden a linux box
 (2) How to do failover.

 for (1), be sure telnet, ftp and any other service you do not
 need is off.  Move standard services to non-standard ports,
 especially web and ssh.  Do not run a name server on the box.

 For (2): You need to have a secondary box that runs a mirror
 copy of Asterisk and mysql and pretty much has everything
 else configured the same.  mysql should be replicated to the
 second box.  You then run a program on the second box that
 pings the first box.  If the first box fails the second takes
 over the first box's IP and runs with it.  There are
 heartbeat programs that can help out with this.

 W


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Re: [asterisk-users] Intercom mode on Polycom and/or SPA9xx

[Tom Vile]

Have you tried using the page() feature.  It has a d option for full
duplex but if you use it without the d it will be one way..

   *  d - full duplex audio
   * q - quiet, do not play beep to caller

exten = 305,1,Page(LOCAL/[EMAIL PROTECTED]|d)


On 7/11/06, C F [EMAIL PROTECTED] wrote:

Made me laugh a bit, but no, I can't it's a discusting looking phone,
and full duplex speaker IS a requirment.

On 7/11/06, Brian Vincent (C) [EMAIL PROTECTED] wrote:

  At the moment I'm using for the Polycom ALERTINFO to a customized ring
  that auto answers, and for the Sipura spa941 SipAddHeaders that also
  autoanswers however they both do 2 way audio, is there anyway that it
  can be configured to 1 way audio?

 Really dumb solution - can you use Polycom 301's?  There's no mic built
 into it, so you need to pick up the handset for the calling party to
 hear you.  For you this might be a feature.

 ---
 Brian Vincent
 Copper Mountain Telecom
 [EMAIL PROTECTED]

 
__

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Re: [asterisk-users] Intercom mode on Polycom and/or SPA9xx

[Tom Vile]

Excuse me for mis-understanding how you talk dude.

1. I have to be able to switch to full duplex in the callees will

Anyone else understand that?

On 7/11/06, C F [EMAIL PROTECTED] wrote:

I already said in the original post why not:
1. I have to be able to switch to full duplex in the callees will
2. It uses meetme
3. I need to be able to complete the xfer.

Thank You

On 7/11/06, Tom Vile [EMAIL PROTECTED] wrote:
 Have you tried using the page() feature.  It has a d option for full
 duplex but if you use it without the d it will be one way..

 *  d - full duplex audio
 * q - quiet, do not play beep to caller

 exten = 305,1,Page(LOCAL/[EMAIL PROTECTED]|d)


 On 7/11/06, C F [EMAIL PROTECTED] wrote:
  Made me laugh a bit, but no, I can't it's a discusting looking phone,
  and full duplex speaker IS a requirment.
 
  On 7/11/06, Brian Vincent (C) [EMAIL PROTECTED] wrote:
  
At the moment I'm using for the Polycom ALERTINFO to a customized ring
that auto answers, and for the Sipura spa941 SipAddHeaders that also
autoanswers however they both do 2 way audio, is there anyway that it
can be configured to 1 way audio?
  
   Really dumb solution - can you use Polycom 301's?  There's no mic built
   into it, so you need to pick up the handset for the calling party to
   hear you.  For you this might be a feature.
  
   ---
   Brian Vincent
   Copper Mountain Telecom
   [EMAIL PROTECTED]
  
   
__
  
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only for the use of the intended recipient(s),
   are confidential, and may be privileged. If you are not the intended 
recipient, you are hereby notified that any review,
   retransmission, conversion to hard copy, copying, circulation or other 
use of this message and any attachments is strictly
   prohibited. If you are not the intended recipient, please notify the 
sender immediately by return e-mail, and delete this
   message and any attachments from your system. Thank you.
   
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Re: [asterisk-users] Re: Metermaid phone compatibility

[Tom Vile]

I can confirm that the 1.2.7.1 patch works with 1.2.9.1 as well.

On 7/10/06, Steven [EMAIL PROTECTED] wrote:

The metermaid changes in head are very different, but there is a working 
1.2.7.1 patch in the bug tracker.
http://bugs.digium.com/view.php?id=5779

I believe that the 1.2.7.1 patch also works with 1.2.9.1.

--
--
Steven

http://www.glimasoutheast.org



Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 Interesting... will this patch (metermaid) work with 1.2.7 asterisk?

 On 7/7/06, shadowym [EMAIL PROTECTED] wrote:

 I have been experimenting with the new metermaid application that allows
 phones to monitor the status of a parked call using BLF.  Does anyone know
 what BLF feature the phone needs to support to make this work.  Is it
 basically the same as the Bristuff Devstate()?  Anyone know which phones do
 and do not support this (metermaid, not Bristuff)?  Of course SNOM seems to
 be the main one but there must be others.  Sounds like perhaps the Polycoms
 work with it as well.

 The reason I am asking is that I have an Aastra 9133i with v1.4 firmware and
 I can't get it to work with metermaid or Devstate().  Aastra tech support
 phoned me about my Bristuff Devstate() question to them and indicated their
 phone does not support that with current firmware but they are looking at it
 for a future release.  That answers my Devstate() question.

 The phone/firmware supports BLF monitoring of SIP extensions just fine.
 Someone on the bug issue in question
 http://bugs.digium.com/view.php?id=5779 stated they had their Aastra working
 with metermaid just fine so I am wondering if I am missing something here.
 The 480i and 9133i are both pretty much the same in terms of BLF support so
 which model I have shouldn't matter.Still scratching my head over the
 person who posted that and I don't know their email to confirm.

 Maybe he is heredimitripietro??

 I am pretty sure I have it set up right.  My GXP2000 seems to work with
 metermaid ok but show hints only shows the GXP2000 monitoring the call
 parking extension (701).  Ie. It only shows one extension monitoring and
 since the GXP2000 is working that must be the one.  I have a second
 extension configured and it is the Aastra 9133i.  Of course I tried a few
 different settings in the Aastra GUI and messed around with the Asterisk
 config.
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Re: [asterisk-users] CallerID

[Tom Vile]

and what TDM card are you using and what does your zapata.conf file look like.

On 7/8/06, Ryder Brook [EMAIL PROTECTED] wrote:

Hope someone call help me .

I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.

I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's not clear if I have to activate for CallerID in the
dialplan. The voicemail keeps saying  call from an unknown caller  etc.
Eventually, i would like to pass on the callerID and name to a pager, if the
call is not picked up, at the extension, after hanging up the call.

Thanks,
braman


Ryder Brook Pediatrics
P.O.Box 608
Morrisville, VT 05661


 
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+
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Re: [asterisk-users] Metermaid phone compatibility

[Tom Vile]

It is working for my and my Snom phones.

On 7/7/06, Matt [EMAIL PROTECTED] wrote:

Interesting... will this patch (metermaid) work with 1.2.7 asterisk?

On 7/7/06, shadowym [EMAIL PROTECTED] wrote:

 I have been experimenting with the new metermaid application that allows
 phones to monitor the status of a parked call using BLF.  Does anyone know
 what BLF feature the phone needs to support to make this work.  Is it
 basically the same as the Bristuff Devstate()?  Anyone know which phones do
 and do not support this (metermaid, not Bristuff)?  Of course SNOM seems to
 be the main one but there must be others.  Sounds like perhaps the Polycoms
 work with it as well.

 The reason I am asking is that I have an Aastra 9133i with v1.4 firmware and
 I can't get it to work with metermaid or Devstate().  Aastra tech support
 phoned me about my Bristuff Devstate() question to them and indicated their
 phone does not support that with current firmware but they are looking at it
 for a future release.  That answers my Devstate() question.

 The phone/firmware supports BLF monitoring of SIP extensions just fine.
 Someone on the bug issue in question
 http://bugs.digium.com/view.php?id=5779 stated they had their Aastra working
 with metermaid just fine so I am wondering if I am missing something here.
 The 480i and 9133i are both pretty much the same in terms of BLF support so
 which model I have shouldn't matter.Still scratching my head over the
 person who posted that and I don't know their email to confirm.

 Maybe he is heredimitripietro??

 I am pretty sure I have it set up right.  My GXP2000 seems to work with
 metermaid ok but show hints only shows the GXP2000 monitoring the call
 parking extension (701).  Ie. It only shows one extension monitoring and
 since the GXP2000 is working that must be the one.  I have a second
 extension configured and it is the Aastra 9133i.  Of course I tried a few
 different settings in the Aastra GUI and messed around with the Asterisk
 config.
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Re: [asterisk-users] Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link)

[Tom Vile]

This was written for use with AAH but should work for you as well.

http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#722Polycom

On 7/6/06, Crazy Boy [EMAIL PROTECTED] wrote:

Hi Friends,

I am using Polycom IP 501 hard phone. Now, I want to confiugre my Polycom IP
501 phone with my Asterisk. I am unable to find any good link or tutorial
for this. Please give a good link to configure my Polycom IP 501 phone with
my Asterisk. Looking forward for your response.

Thank you.

Regards,
Chandra.


 
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Re: [asterisk-users] for you guys setting up customer offices...

[Tom Vile]

Snom 360

On 7/6/06, Fabio [EMAIL PROTECTED] wrote:

Polycom 300, SPA 841/941 (841 is out of market...), pap2 (2 x fxs ata)

Fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Shaun
Enviado el: Jueves, 06 de Julio de 2006 04:45 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] for you guys setting up customer offices...


What brand/model phones are you using.

--

~Shaun



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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

[Tom Vile]

You have to lower the registration interval in the phones to under a
minute otherwise the NAT hole closes and no calls come in.

Polycom has said that they are going to be putting in a keep alive in
the firmware at some point.

On 6/28/06, Von L. [EMAIL PROTECTED] wrote:

Hello,

Here is a breakdown of the issue I am experiencing. I have three remote
employees, in various states, who have Polycom 501 phones. They are
unable to receive incoming calls after a few minutes of the phones being
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.

They each have NAT routers, and I have triple checked that they have
opened/forwarded the correct ports, basically 5060-3 UDP. Once they
plug the phone it (power and ethernet) I see on the CLI console of the
asterisk server that the phones register:

Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on
bell (pid = 3652)
nell*CLI
Verbosity is at least 10
-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600

Here is the top part of my sip.conf

;_
;sip.conf
;_

[general]
port=5060
bindaddr=0.0.0.0
externip=XXX.XXX.XXX.XXX
localnet=XXX.XXX.XXX.XXX/255.255.255.248
canreinvite=no
tos=reliability
srvlookup=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
ignoreregexpire=yes

I know it has something to do with the NAT because if I plug my Polycom
directly into my cable modem, thus making it sit on the Internet and
have a real IP, everything works just fine.

I am curious what I am missing.

Thanks.

Von L.
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www.baldwintechsolutions.com
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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

[Tom Vile]

FYI, when we had NAT routers at both locations setting qualify=yes
did not work.

On 6/28/06, Michiel van Baak [EMAIL PROTECTED] wrote:

On 12:04, Wed 28 Jun 06, Von L. wrote:
 Hello,
 ;_
 ;sip.conf
 ;_

 [general]
 port=5060
 bindaddr=0.0.0.0
 externip=XXX.XXX.XXX.XXX
 localnet=XXX.XXX.XXX.XXX/255.255.255.248
 canreinvite=no
 tos=reliability
 srvlookup=yes
 disallow=all
 allow=ulaw
 dtmfmode=rfc2833
 nat=yes
 ignoreregexpire=yes

Show us one of the phone entries.
Basically check if the following is set there:
nat=yes
qualify=yes

The qualify=yes will send packets so the nat states stay open.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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

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Re: [Asterisk-Users] Mail loop?

[Tom Vile]

Me too.

On 6/27/06, Steven Ringwald [EMAIL PROTECTED] wrote:

Mike Fedyk wrote:
 Is anyone else getting messages from the lists.digium.com mail server
 with errors about a mail loop?

 I've been getting this for the last few weeks, but I don't have any
 list software on my server.  Any ideas?


Yep. I have been getting them quite steadily today. Looks like every
email I ever sent to the list is coming back to me now.

Steve

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Re: [Asterisk-Users] Can I get caller id passed to a phone connected to a Supura 2100?

[Tom Vile]

Probably the ring voltage is to low for the phone.  Try turning it up a bit.

On 6/24/06, Jonathan Attwood [EMAIL PROTECTED] wrote:

Does the Sipura web interface on the info page reveal that the spa2100
is successfully receiving CLID?

My SPA2100 passes CLID from asterisk to the connected phone without problem.

On 23/06/06, Jim Lynch [EMAIL PROTECTED] wrote:
 I have a Uniden wireless phone connected into Linksys/Supura 2100.  It
 works well, except I never see any caller ID information displayed on
 the phone.  Is that a setting in the 2100 that I'm missing, or is it an
 Asterisk setting or isn't it possible?

 Thanks,
 Jim.
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Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls

[Tom Vile]

Also check the dialplan on the ATA as well.  Maybe its the way she is
dialing the number that is causing the issue.

On 6/21/06, Leah Newmark [EMAIL PROTECTED] wrote:

Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!

Anyway, she *is* able to receive calls. She gets a fast busy when trying
to dial anything.

I know we had her do speed tests on her DSL the end of last year but I
don't remember the outcome, but that was for quality issues, so I don't
think it has to do with this problem per se.

Any ideas of what to test or look at?

Thanks!

LN

Message: 18
Date: Tue, 20 Jun 2006 00:12:51 +0200
From: lenz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] User Loses Ability to Make Outgoing
   Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; delsp=yes;
   charset=iso-8859-15


Hello Leah,
it may be the quality of her link degrading - it happens easily with ADSL.
which error does she get? and she cannot receive calls at the same time,
right?
l.




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Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls

[Tom Vile]

Again, Also check the dialplan on the ATA as well.  Maybe its the way she is
dialing the number that is causing the issue.  If its sporadic then
that may be the issue.  Does only happen to certain numbers she dials
or does it happen with the same numbers she dials randomly?

On 6/21/06, Leah Newmark [EMAIL PROTECTED] wrote:

That's not the problem. The contexts are all fine, and the problem fixes
itself when it feels like it. I am almost positive it's her connection.
The asterisk coding on my end is fine. She has the same setup as the
other 20 employees and they all work fine.

She was running tests on dsltools.com. Is there any specific test I need
to do to diagnose the problem?

[EMAIL PROTECTED] wrote:

Message: 14
Date: Wed, 21 Jun 2006 10:28:50 -0600
From: Michael Welter [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing
Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Look in the sip.conf (or whatever) and make sure the context specifies
a context that allows outgoing calls.

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Re: [Asterisk-Users] Asterisk -- BV: Incoming does not work....

[Tom Vile]

Try these settings

context=incoming-bv
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=MyPassword
type=user
user=XX
username=XX
user=phone
username=XX

On 6/21/06, John Klimek [EMAIL PROTECTED] wrote:

Anybody else able to help...?

On 6/19/06, John Klimek [EMAIL PROTECTED] wrote:
 Ahh, good catch.  I've changed the context to be incoming-bv (to
 match my context in extensions.conf), but I still get the same exact
 phone message...

 Also, is it normal to see REGISTER attempt 1 to
 [EMAIL PROTECTED]@sip.broadvoice.com in the asterisk
 console being repeated every 20 or so seconds?


 On 6/19/06, Tom Vile [EMAIL PROTECTED] wrote:
  your context=incoming does not match your context in extensions.conf 
incoming-bv
 
  On 6/19/06, John Klimek [EMAIL PROTECTED] wrote:
   Asterisk seems to register just fine with BroadVoice (asterisk -r, and
   then sip show registry shows sip.broadvoice.com is Registered)
  
   ...but when I try to call my broadvoice number (from a cell phone), it
   rings one single time and then says The party you are trying to reach
   is not available to take your call.  This doesn't seem to be an
   Asterisk message but seems to be coming from someplace else.
  
   What can I be doing wrong?  I've searched the forums and mailing lists
   and my config seems to be perfectly fine...
  
   Here are my config snippets:
  
   *** SIP.CONF ***
   [sip.broadvoice.com]
   type=friend
   host=sip.broadvoice.com
   fromdomain=sip.broadvoice.com
   fromuser=5703380128
   username=5703380128
   authname=5703380128
   secret=##--- hidden
   insecure=very
   context=incoming
   dtmfmode=inband
   dtmf=inband
   canreinvite=no
   nat=yes
  
   *** EXTENSIONS.CONF ***
  
   [incoming-bv]
   exten = s,1,Answer()
   exten = 2100,1,Answer()
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  www.baldwintechsolutions.com
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Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls

[Tom Vile]

When she gets the fast busy is she registered to the server?

On 6/21/06, Leah Newmark [EMAIL PROTECTED] wrote:

We've tried 2 different ATAs, the same hardware, same setup as everyone
else and both have this problem.
This problem is for any number she dials, whether she dials her voicemal
(in-system) or makes a long distance call. She just gets a fast busy,
and I see no output on the console whatsoever.


[EMAIL PROTECTED] wrote:



Message: 1
Date: Wed, 21 Jun 2006 13:44:58 -0400
From: Tom Vile [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing
   Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
Message-ID:
   [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Also check the dialplan on the ATA as well.  Maybe its the way she is
dialing the number that is causing the issue.






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www.baldwintechsolutions.com
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Re: [Asterisk-Users] Asterisk -- BV: Incoming does not work....

[Tom Vile]

your context=incoming does not match your context in extensions.conf incoming-bv

On 6/19/06, John Klimek [EMAIL PROTECTED] wrote:

Asterisk seems to register just fine with BroadVoice (asterisk -r, and
then sip show registry shows sip.broadvoice.com is Registered)

...but when I try to call my broadvoice number (from a cell phone), it
rings one single time and then says The party you are trying to reach
is not available to take your call.  This doesn't seem to be an
Asterisk message but seems to be coming from someplace else.

What can I be doing wrong?  I've searched the forums and mailing lists
and my config seems to be perfectly fine...

Here are my config snippets:

*** SIP.CONF ***
[sip.broadvoice.com]
type=friend
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=5703380128
username=5703380128
authname=5703380128
secret=##--- hidden
insecure=very
context=incoming
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes

*** EXTENSIONS.CONF ***

[incoming-bv]
exten = s,1,Answer()
exten = 2100,1,Answer()
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Re: [Asterisk-Users] No CID on ZAP

[Tom Vile]

Thanks for sharing that info.

How about sharing your zapata.conf configuration so that someone can
look at it and maybe see if there is a problem.

I'm guessing you want help with this.

On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote:





I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs
line coming into a Digium TDM01B. It appears to not be getting CID at all.
If I hook up a POTS phone to the line CID comes through fine. Inbound and
outbound calls work fine but there is just no CID on inbound for this
channel.The incoming route for the channel is Zaptel Channel 0. No DID or
CID settings applied. My IP routes inbound are providing CID with no issue.





Here is the output from the log when a call is coming in:



-- Starting simple switch on 'Zap/1-1'

-- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new
stack

-- Executing Set(Zap/1-1, DID=s) in new stack

-- Executing NoOp(Zap/1-1, DID is now s) in new stack

-- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack

-- Goto (from-zaptel,s,7)

-- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack

-- Executing Set(Zap/1-1, CHAN=1-1) in new stack

-- Executing Set(Zap/1-1, CHAN=1) in new stack

-- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack

-- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new
stack

-- Executing Goto(Zap/1-1, ext-did|s|1) in new stack

-- Goto (ext-did,s,1)

-- Executing Set(Zap/1-1, FROM_DID=s) in new stack

-- Executing Goto(Zap/1-1, ext-local|200|1) in new stack

-- Goto (ext-local,200,1)

-- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack

-- Executing Macro(Zap/1-1, user-callerid) in new stack

-- Executing GotoIf(Zap/1-1, 0?report) in new stack

-- Executing GotoIf(Zap/1-1, 0?start) in new stack

-- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack

-- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack

-- Executing Set(Zap/1-1, AMPUSER=) in new stack

-- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack

-- Executing GotoIf(Zap/1-1, 1?report) in new stack

-- Goto (macro-user-callerid,s,9)

-- Executing NoOp(Zap/1-1, Using CallerID  ) in new stack

-- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack

-- Executing Set(Zap/1-1, VMBOX=200) in new stack

-- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack

-- Executing Set(Zap/1-1, CFUEXT=) in new stack

-- Executing Set(Zap/1-1, RT=25) in new stack

-- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack

-- Executing GotoIf(Zap/1-1, 0  0?2:4) in new stack

-- Goto (macro-record-enable,s,4)

-- Executing AGI(Zap/1-1,
recordingcheck|20060609-095557|1149864957.408) in new
stack

-- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck

  recordingcheck|20060609-095557|1149864957.408: Inbound
recording not enabled

-- AGI Script recordingcheck completed, returning 0

-- Executing NoOp(Zap/1-1, No recording needed) in new stack

-- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack

-- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack

-- Executing AGI(Zap/1-1, dialparties.agi) in new stack

-- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi

dialparties.agi: Starting New Dialparties.agi

--  dialparties.agi: priority is 1  dialparties.agi: Caller ID name is
'unknown' number is 'unknown'

 dialparties.agi: Methodology of ring is  'none'

--  dialparties.agi: Added extension 200 to extension map

--  dialparties.agi: Extension 200 cf is disabled

--  dialparties.agi: Extension 200 do not disturb is disabled

  == Parsing '/etc/asterisk/manager.conf': Found

  == Manager 'admin' logged on from 127.0.0.1

  == Manager 'admin' logged off from 127.0.0.1

--  dialparties.agi: Checking CW and CFB status for extension 200

--  dialparties.agi: DbSet CALLTRACE/200 to unknown

-- AGI Script dialparties.agi completed, returning 0

-- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack



Any help would be appreciated.







Thanks



Curt
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Re: [Asterisk-Users] Plainvoip problem.

[Tom Vile]

Do you have the g729 codec?

On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:

When calling through Plainvoip from my Asterisk at Home box I get the
following log entries.

Jun  8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert #
Jun  8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by
66.199.240.2 (format g729)
Jun  8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729
Jun  8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing
Jun  8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped
sounds
Jun  8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered
SIP/503-6d4c
Jun  8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256

What I hear on the phone is one ring and then nothing.

This has only been in the past few days.

Has anybody else had a problem like this?

--
Henry J. Cobb
http://www.io.com/~hcobb/

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Re: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf

[Tom Vile]

you put custom code in the _custom.conf files not in the
_addtional.conf files.  You can also add the changes directly to the
Mysql database as well and they will be included in the
_addtional.conf files.

On 6/8/06, Lachek Butalek [EMAIL PROTECTED] wrote:

Figuring I knew what I was doing (I didn't - surprise) I added a
totally unnecessary line in /etc/asterisk/extensions_additional.conf a
couple of days ago. Troubleshooting a dialing rule issue, I'm now
realizing that FreePBX is updating its database with the new settings
but is not rewriting/updating extensions_additional.conf with the
changes I'm making.

I've tried renaming the file, changing its ownership, changing its
permissions, restarting the portal, all without any success. Web
resources on this issue claim the opposite problem - that custom
changes to extensions_additional.conf will be automatically rewritten
every time FreePBX/AMP is updated. If that was true, I'd be done -
unfortunately, it seems this is not the case.

I really don't want to reinstall FreePBX and redo my entire
configuration again... :(

Any assistance would be greatly appreciated.
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www.baldwintechsolutions.com
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Re: [Asterisk-Users] Phone recommendations?

[Tom Vile]

Search the list. This question gets ask almost every week.

I prefer:

Snom
Polycom
Cisco

What not too pricey?

On 6/8/06, Derek [EMAIL PROTECTED] wrote:

Hi All,

I'm looking for a good voip hardphone that has a decent set of the
regular features (conference, 2 lines, etc) thats reliable, has decent
quality, and isn't too pricey. Does anyone have any suggestions?

Thanks in advance.
Derek

--
Derek Fedel


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Re: [Asterisk-Users] Re: GXP-2000 (steer clear)

[Tom Vile]

Anyone try out the Snom 300 phone yet?  Seems like a decent price.

On 6/7/06, Matthias Fechner [EMAIL PROTECTED] wrote:

Hello Mimmus,

* Mimmus [EMAIL PROTECTED] [07-06-06 17:20]:
 Yes

good to known.
I played with the idea to buy one of these.

You would suggest GrandStream then?

Best regards,
Matthias
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Re: [Asterisk-Users] DTMF feedthru again...

[Tom Vile]

try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:

Ok trying this again... is there anyone using the SPA-3000 with * 
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!

When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.

I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.

What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.

I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.

How can I fix this???

Doug


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



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Re: [Asterisk-Users] DTMF feedthru again...

[Tom Vile]

Using AVT in my sipura with above settings and it work fine going out
the PSTN.  There was an issue a while back with an older version of
Asterisk with one of my providers but it has been fine since the
upgrade.  I also use ulaw for calls.

On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:

Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem in my book. It precludes me from using any DTMF over PSTN with *
at this point.

Any further help or explanation would be appreciated.

On Tue, 6 Jun 2006, Tom Vile wrote:

 try setting dtmf playback length to .5 in the admin section of the
 Sipura and try again.

 On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
  Ok trying this again... is there anyone using the SPA-3000 with * 
  I am not sure if this is a specific problem to it or not. This is
  something I really need to fix!!!
 
  When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
  access (reliably) DTMF menus at the called party, after call completion.
  Dialing DTMF is fine.
 
  I checked by calling myself. Listening to either end on a completed call,
  and pressing a DTMF button on the opposing phone results in an audible
  click and very little if any audible DTMF energy being heard.
 
  What is muting the DTMF??? Does * have anything to do with this? I am
  not using any dial flags.
 
  I tried 'inband' in all places with no difference. At one point this
  seemed like a * feature problem and I thought removing dial flags fixed
  it but that does not now seem to be the case.
 
  How can I fix this???
 
  Doug
 
  
  *  Doug Crompton   *
  *  Richboro, PA 18954  *
  *  215-431-6307*
  *  *
  * [EMAIL PROTECTED]*
  * http://www.crompton.com  *
  
 
 
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 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
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 Fax: 518-631-2856
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 deserve neither liberty nor safety.  -- Ben Franklin (1759)


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



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www.baldwintechsolutions.com
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Re: [Asterisk-Users] DTMF feedthru again...

[Tom Vile]

AVT is the DTMF setting.  Yes, I hear the tones fine.  Play with the
dtmf playback length and adjust it, also play with the dtmf playback
level its default is .16 I believe.

On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:

AVT??? I have ulaw allowed (only) - When you call your cell via
pstn/spa-3000/* and listen on both while pressing dtmf do you hear good
clean tones of enough duration to allow detection, in both directions?

Do you access DTMF required services over pstn, like banking, vm, etc
from local * system?

Doug

On Tue, 6 Jun 2006, Tom Vile wrote:

 Using AVT in my sipura with above settings and it work fine going out
 the PSTN.  There was an issue a while back with an older version of
 Asterisk with one of my providers but it has been fine since the
 upgrade.  I also use ulaw for calls.

 On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
  Tried that makes no difference. Did it for you? What DMF method(s) are you
  using. Looking at a goggle search yields lots of talk on this but no real
  solution. Apparently there is an rfc2833 issue and * is working on it???
  Also it appears the codec used might be an issue. This is a serious
  problem in my book. It precludes me from using any DTMF over PSTN with *
  at this point.
 
  Any further help or explanation would be appreciated.
 
  On Tue, 6 Jun 2006, Tom Vile wrote:
 
   try setting dtmf playback length to .5 in the admin section of the
   Sipura and try again.
  
   On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
Ok trying this again... is there anyone using the SPA-3000 with * 
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
   
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
   
I checked by calling myself. Listening to either end on a completed 
call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
   
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
   
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
   
How can I fix this???
   
Doug
   

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

   
   
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   Baldwin Technology Solutions, Inc
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   www.baldwintechsolutions.com
   Phone: 518-631-2855 x205
   Fax: 518-631-2856
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  Those that sacrifice essential liberty to obtain a little temporary safety
   deserve neither liberty nor safety.  -- Ben Franklin (1759)
 
  
  *  Doug Crompton   *
  *  Richboro, PA 18954  *
  *  215-431-6307*
  *  *
  * [EMAIL PROTECTED]*
  * http://www.crompton.com  *
  
 
 
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 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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 deserve neither liberty nor safety.  -- Ben Franklin (1759)


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



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Re: [Asterisk-Users] DTMF feedthru again...

[Tom Vile]

But access to voicemail in Asterisk will not work with inband.

On 6/6/06, Matt [EMAIL PROTECTED] wrote:

I've never had this problem with the SPA-2002s... but you could always
set DTMF to 'inband' rather then RFC.. that will cause whatever goes
out of the phone to go right across with no transcoding.

On 6/6/06, Mike Lynchfield [EMAIL PROTECTED] wrote:
 in Fact we saw similar problems with all sipura products.

 We think its a default value thats not quite right for the north american
 market, these units are built and tested in asia mostly.

 one simple test to check it out is call this number
 www.nextwavetitaniumplus.com Toll-Free Account Information Line:
 888-252-9535
 it just seemd that even the cisco is not passing the dtmf ..

 Can anyone confirm ?




 On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
  Also to expand on this... when listening to opposing phone in a connected
  call over PSTN you hear a click followed by a very short burst of DTMF
  audible energy. Same in both directions.
 
  I can't be the only one having this problem!
 
  Doug
 
  On Tue, 6 Jun 2006, Tom Vile wrote:
 
   try setting dtmf playback length to .5 in the admin section of the
   Sipura and try again.
  
   On 6/6/06, Doug Crompton  [EMAIL PROTECTED] wrote:
Ok trying this again... is there anyone using the SPA-3000 with * 
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
   
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call
 completion.
Dialing DTMF is fine.
   
I checked by calling myself. Listening to either end on a completed
 call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
   
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
   
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags
 fixed
it but that does not now seem to be the case.
   
How can I fix this???
   
Doug
   

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

   
   
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   --
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   Baldwin Technology Solutions, Inc
   Consulting - Web Design - VoIP Telephony
   www.baldwintechsolutions.com
   Phone: 518-631-2855 x205
   Fax: 518-631-2856
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  Those that sacrifice essential liberty to obtain a little temporary
 safety
  deserve neither liberty nor safety.  -- Ben Franklin (1759)
 
  
  *  Doug Crompton   *
  *  Richboro, PA 18954  *
  *  215-431-6307*
  *  *
  * [EMAIL PROTECTED]*
  * http://www.crompton.com  *
  
 
 
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Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] DTMF feedthru again...

[Tom Vile]

I meant info not inband.  Sorry.

On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:

What do you mean you cannot access VM?

I am totally inband here on SPA3000 to fix the DTMF feedthru problem and
there is no problem with * VM. I can access it from local analog phone or
I can call in and '*' it to get PW prompt.

Doug


On Tue, 6 Jun 2006, Tom Vile wrote:

 But access to voicemail in Asterisk will not work with inband.



 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
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Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety.  -- Ben Franklin (1759)


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



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Baldwin Technology Solutions, Inc
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www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] DID in Houston 713?

[Tom Vile]

I've been using Viatalk for 2 months and have been down 3x.  They had
a misconfiguration on there end that they would not fix until quite a
few of us started having the issue.  Tech support has been pretty bad
for the most part and unwilling to cooperate.  So I would not suggest
them at this point.

On 6/2/06, George A. Roberts IV [EMAIL PROTECTED] wrote:

ViaTalk shows that they have service in Houston and they support porting.
http://www.viatalk.com/

Been using them for about 6 months with our * box and they've been rock
solid.

Regards,

George A. Roberts IV
President and CEO, Interjuncture Corp.
http://www.interjuncture.com/

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, June 01, 2006 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] DID in Houston 713?

Does anyone on-list know of a serice provider that can provide DIDs in
713-861-? I'd like to port my ATT POTS lines to an IP based service
into my Asterisk box.

Michael



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Re: [Asterisk-Users] DID from Latvia?

[Tom Vile]

try voxbone.com

On 6/2/06, David K Parker [EMAIL PROTECTED] wrote:

Does anyone know of a good VOIP provider that I can obtain a DID from
Latvia? I live in the US but have a friend there.

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Re: [Asterisk-Users] DID from Latvia?

[Tom Vile]

You only mentioned getting a DID not placing calls out through them
and that is why I mentioned them.  It is a very useful service in that
I can have DID's in other countries and route them to my Asterisk
server.  There are a number of outgoing providers to many countries
that you can use but you will pay per minute charges.

On 6/2/06, David K Parker [EMAIL PROTECTED] wrote:


Wow, I went through and setup an acoount but couldn't figure out why I could
only get a fast busy signal when dialing, then, checking the FAQs I noticed
this very important detail, you cannot place outgoing calls over Voxbone.
How messed up is that? They don't bother to mention this fact in there
service offerings.

http://www.voxbone.com/faq-services.jsf#serv8

Can I place outbound calls via the Voxbone network?
= No. We are specialized in VoIP origination services (from PSTN to VoIP)
and we do not provide termination. In other words you can only receive calls
from the Voxbone network.


On 6/2/06, Tom Vile [EMAIL PROTECTED] wrote:
 try voxbone.com

 On 6/2/06, David K Parker [EMAIL PROTECTED] wrote:
  Does anyone know of a good VOIP provider that I can obtain a DID from
  Latvia? I live in the US but have a friend there.
 
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Fax: 518-631-2856
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Re: [Asterisk-Users] Openion on Sipura SPA-2100

[Tom Vile]

Sipura 3000 or the Digium TDM03B

On 6/1/06, Crazy Boy [EMAIL PROTECTED] wrote:

Hi,

As you said, May I know the correct Digium or Sipura product model
(Sipura-3102 or Digium?), which is suitable to my requirements?

Thank you.

Regards,
Chandramouli


Martin Joseph [EMAIL PROTECTED] wrote:


On May 31, 2006, at 10:32 PM, Crazy Boy wrote:

 Hi Friends,

 I have successfully implemented Intercom, Voicemail and International
 dialing using Asterisk. Now I want to connect my PSTN Lines to
 Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk.
 For this, I want to use Sipura SPA-2100. Is my decession is correct or
 not? Is there any disadvantages with this Sipura SPA-2100? Please tell
 me.

 The SPA-2100 is an FXS, which allows the connection of phone handsets
to your asterisk server.

If you want to hook up phone lines you need an FXO.

Marty

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Re: [Asterisk-Users] Upgrade ONLY asterisk from an AAH install

[Tom Vile]

The upgrade script will not work.  There are detailed instructions at
the AAH Wiki on voip-info.org

Upgrading AAH is not painful at all, as long as you know how to update
the individual pieces yourself.

On 5/31/06, Dan Elder [EMAIL PROTECTED] wrote:

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH
to upgrade only the asterisk binaries? Doug has chimed in a few times saying
'upgrade' when I post problems, but Aah makes this really painful. I'm using
AAH 2.0  am fighting a number of 'bugs' that only seem to be manifesting in
my installation. Can I safely upgrade just asterisk and not any of the other
AAH components without borking the whole install? It's a production machine,
so I'm extremely timid to make any major changes on the box  accidentally
take a chunk of users offline.

Thanks as always!

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Re: [Asterisk-Users] Upgrade ONLY asterisk from an AAH install

[Tom Vile]

http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+2#297UpdatingAsteriskmanuallypreferredsoyo

On 5/31/06, David K Parker [EMAIL PROTECTED] wrote:

hmm, I couldn't find any upgrade procedures there, only initial install


On 5/31/06, Tom Vile [EMAIL PROTECTED]  wrote:
 The upgrade script will not work.  There are detailed instructions at
 the AAH Wiki on voip-info.org

 Upgrading AAH is not painful at all, as long as you know how to update
 the individual pieces yourself.

 On 5/31/06, Dan Elder  [EMAIL PROTECTED] wrote:
  Hey all, is it safe to run the asterisk-update.sh script that comes with
AAH
  to upgrade only the asterisk binaries? Doug has chimed in a few times
saying
  'upgrade' when I post problems, but Aah makes this really painful. I'm
using
  AAH 2.0  am fighting a number of 'bugs' that only seem to be
manifesting in
  my installation. Can I safely upgrade just asterisk and not any of the
other
  AAH components without borking the whole install? It's a production
machine,
  so I'm extremely timid to make any major changes on the box 
accidentally
  take a chunk of users offline.
 
  Thanks as always!
 
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 --
 Tom Vile
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 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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Re: [Asterisk-Users] Now that Nufone is dead...

[Tom Vile]

Then you are a luck one aren't you.  Haven't had my 800 number for
over a month now but they still bill you for having the number.
Interesting.

On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1


On 23 May 2006, at 15:48, Carlos Chavez wrote:

  Now that Nufone is dead, what are other providers of 800
 numbers that
 work with Asterisk?

Nufone is not dead, works perfectly fine for me.

jens



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Version: GnuPG v1.4.1 (Darwin)

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+U9WV0uDc/qD2uhr5AmyAfw=
=rfHQ
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Now that Nufone is dead...

[Tom Vile]

$2.50 p/month for 800 DID.

On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

They bill you for having the 800 number? I thought they only did that
for Michigan DIDs. They only bill my actual call time.

jens

On 23 May 2006, at 16:54, Tom Vile wrote:

 Then you are a luck one aren't you.  Haven't had my 800 number for
 over a month now but they still bill you for having the number.
 Interesting.

 On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1


 On 23 May 2006, at 15:48, Carlos Chavez wrote:

   Now that Nufone is dead, what are other providers of 800
  numbers that
  work with Asterisk?

 Nufone is not dead, works perfectly fine for me.

 jens



 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.1 (Darwin)

 iD8DBQFEcybORAx5nvEhZLIRAnAoAJwJ0Ig4EUdrfw1RhTe8ULxzzq3dQQCfesYc
 +U9WV0uDc/qD2uhr5AmyAfw=
 =rfHQ
 -END PGP SIGNATURE-
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 --
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 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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-BEGIN PGP SIGNATURE-
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wyuiw+R12uRQkTp0ZGZTEF0=
=b0Jk
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Now that Nufone is dead...

[Tom Vile]

Definately adds stress to your life when a provider disconnects you
but I can see that they are trying to square things away with there
customers and it looks like good changes are in the works.

On 5/23/06, Manny A. Wise [EMAIL PROTECTED] wrote:

I never liked Jeremy, having that out of the way, :)

What happen to him can happen to ANYONE!

It happened to Broadvoice big time Also Vonage!!.. but they are more
prepared to deal with the root cause!! They have more resources!! And more
MONEY!!! It has nothing to do with reputation!!!

Don't spit out without facts!!!

Manny

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, May 23, 2006 8:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Now that Nufone is dead...
I think the point everyone is making is that no reputable company
would have had this happen.  Can you see Vonage losing all their DIDs?
 No!   NuFone clearly did something that screwed their contract with
their CLEC...
On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 Here's one for all the naysayers: I only sent an email to NuFone
 accounting to inquire about that $2.50/month fee and they're falling
 over themselves to not only get all my questions answered but to also
 helping me getting my account set up in the most economical way for
 me after their upstream provider problems. Proves me right for
 sticking with them.
 jens
 On 23 May 2006, at 21:41, Jens Vagelpohl wrote:

  -BEGIN PGP SIGNED MESSAGE-
  Hash: SHA1
 
  I see it now on the FAQ, but this must be a new thing. I paid $50
  in December 2004 and still have over $39 (yes, I don't use it
  often). If I remember correctly the 800 DIDs were advertised as
  free of monthly fees, call fees only.
 
  jens
 
 
  On 23 May 2006, at 20:13, Tom Vile wrote:
 
  $2.50 p/month for 800 DID.
 
  On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
  -BEGIN PGP SIGNED MESSAGE-
  Hash: SHA1
 
  They bill you for having the 800 number? I thought they only did
  that
  for Michigan DIDs. They only bill my actual call time.
 
  jens
 
  On 23 May 2006, at 16:54, Tom Vile wrote:
 
   Then you are a luck one aren't you.  Haven't had my 800 number for
   over a month now but they still bill you for having the number.
   Interesting.
  
   On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
   -BEGIN PGP SIGNED MESSAGE-
   Hash: SHA1
  
  
   On 23 May 2006, at 15:48, Carlos Chavez wrote:
  
 Now that Nufone is dead, what are other providers of 800
numbers that
work with Asterisk?
  
   Nufone is not dead, works perfectly fine for me.
  
   jens

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Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Confused !

[Tom Vile]

I though g723 used 5.43 KBps and g729 was 5.91 KBps

On 5/16/06, Jon Weisman [EMAIL PROTECTED] wrote:


 Your restriction is what the service provider allows. Most (that I've
 used)
 allow g729. I know it uses more bandwidth than g723 but nothing like G711
 (ulaw or alaw) and from my experience, the quality is quite reasonable.

uh...g729 uses the least bandwidth

-jon



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www.baldwintechsolutions.com
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Re: [Asterisk-Users] S100-FX v2 audio quality

[Tom Vile]

The fix I found was to not use them.  They are not ready yet for production use.

On 5/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

Hi all,

I have 2 of these with me and from day 1 im facing issues with the
background noice i have done all sorts of testing but all in vain. the
changinf of poer supply reduced noise to certain extend but still its
very clearly audible.

A good fix would be really helpful to me

Thanks



On 13/05/06, Ben Holt [EMAIL PROTECTED] wrote:
 Darrick Hartman wrote:
  Tom Vile wrote:
  Same problem with audio quality.  Got rid of them.  Also the context
  line only allowed 12 characters and we need more than that for some
  installations, I didn't want to have to rename 100 contexts to less
  than 12 characters.
  Which audio codecs were you using?  I'm using g729 to connect mine to
  the Asterisk box and don't have any audio problems.  Calls routed out
  over voip and my POTS lines via a TMD400 card both sounds good.  I would
  communicate the problems back to the manufacturer.  They seemed
  interested in feedback.  It is upgradeable via ftp so perhaps the
  context line can be modified by them.

 The audio quality issue I am experiencing is white background noise.  I
 have tried g711u, gsm, and iLBC without any noticeable change.

 I am pretty sure it is not codec related as it is audible even during
 the dial tone or the 'silent' period between dialing and ringing (and of
 course throughout the call). Adjusting the audio gain doesn't seem to
 improve the situation and I have confirmed with the manufacturer that I
 am using the latest firmware.

 - Ben
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www.baldwintechsolutions.com
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Re: [Asterisk-Users] S100-FX v2 audio quality

[Tom Vile]

Same problem with audio quality.  Got rid of them.  Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.

On 5/12/06, Gareth Blades [EMAIL PROTECTED] wrote:

I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.

Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being used.

One minor annoyance is that it is configured with an absolute setting
for the registry interval and does not pay any attention to what
asterisk says. You therefore need to make sure it does not exceed the
asterisk setting otherwise you get continuous asterisk warnings.

Its missing a couple of features though :-

1) You can only configure the callerid number and not the name.

2) Message waiting is not supported.

On Fri, 2006-05-12 at 13:40, Bill Peck wrote:

 On 5/12/06, Ben Holt [EMAIL PROTECTED] wrote:
 Hello,

 In a fit of optimism I recently purchased a X100-FX v2
 (http://x100p.com/products_2.htm) despite the lack of reviews
 I was able
 to find on the device.  The feature set made it hard to
 resist.  I have
 since been experiencing audio quality issues with it.

 Do any other mailing list members have experience with this
 ATA?  If so,
 could you let me know if you are satisfied with its audio
 quality.  At
 this point I don't know if I have a bum unit, a configuration
 problem,
 or am having a typical experience.

 Many thanks,

 I can't even get mine to work. :-(Can you share your configuration
 for it?




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www.baldwintechsolutions.com
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Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

[Tom Vile]

This is what I use:

[ext-paging]
exten = PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
exten = PAGE203,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE203,n,Dial(SIP/203,5)
exten = Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED])
exten = 2030,1,Page(LOCAL/[EMAIL PROTECTED])

if I dial 2030 it will page extension 203.  Change accordingly.  This
works on my Grandstream and Snom phones.


On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:

Asterisk 1.2.7.1 and Zaptel 1.2.5

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, May 11, 2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

What version of Asterisk?

On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:




 I am looking to setup paging using the auto answer feature on the
 Grandstream GXP2000.  I am thinking I will follow the method as described
 here:



 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page



 I will setup the 4th account on the phone to auto answer.



 Does anyone else have a method that works better?  I also looked at the
 allpage AGI written on Voip-Info.  But it seems to dial all extensions,
even
 the ones I don't want to use for Auto Answer.



 I really would like a way to group the phones instead of having them all
 listed in a dial command.



 exten =
 7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]/nLocal/interal
 [EMAIL PROTECTED]|)



 Thanks!
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www.baldwintechsolutions.com
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Fax: 518-631-2856
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www.baldwintechsolutions.com
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Re: [Asterisk-Users] S100-FX v2 audio quality

[Tom Vile]

We did communicate this to the manufacturer and they fixed 1 issue
with bad power supplies.   We tried multiple codecs but it was still
unreliable, so we went back to the IAXy and no issues.  All calls came
in over a PRI.

Did not want to waste to much time on these, maybe we will look again
in the future.



On 5/12/06, Darrick Hartman [EMAIL PROTECTED] wrote:

Tom Vile wrote:
 Same problem with audio quality.  Got rid of them.  Also the context
 line only allowed 12 characters and we need more than that for some
 installations, I didn't want to have to rename 100 contexts to less
 than 12 characters.
Which audio codecs were you using?  I'm using g729 to connect mine to
the Asterisk box and don't have any audio problems.  Calls routed out
over voip and my POTS lines via a TMD400 card both sounds good.  I would
communicate the problems back to the manufacturer.  They seemed
interested in feedback.  It is upgradeable via ftp so perhaps the
context line can be modified by them.

I don't know how well they will perform over time.  I've only had this
unit for 2 weeks.  We were planning on ordering several more for a
client who wants to connect several remote users.  Both will spend
considerable amounts of time on the phone so it has to just work.  For
the advertised feature set, these looked impressive.

Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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www.baldwintechsolutions.com
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Fax: 518-631-2856
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Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

[Tom Vile]

make the dial command like so:

exten = PAGE203,n,Dial(SIP/203SIP/204SIP/205SIP/206,5)

On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:

Thanks.  I like that method.

Do you think if I add all my extensions (say 40 of them) to new Dial
commands after exten = PAGE203,n,Dial(SIP/203,5)

Like this:
exten = PAGE203,n,Dial(SIP/203,5)
exten = PAGE203,n,Dial(SIP/204,5)
exten = PAGE203,n,Dial(SIP/205,5)
exten = PAGE203,n,Dial(SIP/206,5)

will work?

The only think I am not sure about is if a phone doesn't answer in the 5
seconds allowed, it will hang until that 5 seconds is up to parse the next
line.

Also, if it is dialing each phone line by line will there be a delay for all
the phones are dialed and pick up.

Thanks again.

Forrest

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Friday, May 12, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

This is what I use:

[ext-paging]
exten = PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
exten = PAGE203,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE203,n,Dial(SIP/203,5)
exten = Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED])
exten = 2030,1,Page(LOCAL/[EMAIL PROTECTED])

if I dial 2030 it will page extension 203.  Change accordingly.  This
works on my Grandstream and Snom phones.


On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:
 Asterisk 1.2.7.1 and Zaptel 1.2.5

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
 Sent: Thursday, May 11, 2006 6:55 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
GXP2000

 What version of Asterisk?

 On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:
 
 
 
 
  I am looking to setup paging using the auto answer feature on the
  Grandstream GXP2000.  I am thinking I will follow the method as
described
  here:
 
 
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
 
 
 
  I will setup the 4th account on the phone to auto answer.
 
 
 
  Does anyone else have a method that works better?  I also looked at the
  allpage AGI written on Voip-Info.  But it seems to dial all extensions,
 even
  the ones I don't want to use for Auto Answer.
 
 
 
  I really would like a way to group the phones instead of having them all
  listed in a dial command.
 
 
 
  exten =
  7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]/nLocal/interal
  [EMAIL PROTECTED]|)
 
 
 
  Thanks!
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  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 


 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

[Tom Vile]

What version of Asterisk?

On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:





I am looking to setup paging using the auto answer feature on the
Grandstream GXP2000.  I am thinking I will follow the method as described
here:



http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page



I will setup the 4th account on the phone to auto answer.



Does anyone else have a method that works better?  I also looked at the
allpage AGI written on Voip-Info.  But it seems to dial all extensions, even
the ones I don't want to use for Auto Answer.



I really would like a way to group the phones instead of having them all
listed in a dial command.



exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]/nLocal/interal
[EMAIL PROTECTED]|)



Thanks!
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] regarding freepbx

[Tom Vile]

If you read the documentation you would have known that this is not
the case in previous versions and in the latest.  Custom code goes in
the _custom.conf files and they will not be overwritten.  RTFM

On 5/9/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:

Emmo ather wrote:

 In older version of freebpx if you write somethng manually in the
 configuration files it was flushed by amp, i.e. you can configure it
 through the interface only. Is this this thing still present in freepbx?

Why don't you ask that on a FreePBX support list?
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] [EMAIL PROTECTED] Memory Limits

[Tom Vile]

Not sure what he is talking about.  Ask for clarification because it
does not make much sense.

On 5/9/06, scott [EMAIL PROTECTED] wrote:

Hi

I have the latest [EMAIL PROTECTED] installed and everything is working 
perfectly.

I have been told by a collegue that [EMAIL PROTECTED] doesn't use the full 
potential of the machine it is installed on i.e. the CPU  Memory, unless the 
kernel has modified.

He is unsure where he heard this from and I wonderd if anyone had any other 
information about this or knew where I could find some?

Many Thanks
Scott Pinhorne
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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