Re: [Asterisk-Users] Language Problems

[Peter Svensson]

On Sat, 26 Feb 2005, Anton Krall wrote:

> and then Spanish as
> /var/log/asterisk/sounds/sp
> /var/log/asterisk/sounds/sp/phonetic
> /var/log/asterisk/sounds/sp/digit
> /var/log/asterisk/sounds/sp/letters
>
> Now, the normal voices ARE heard in spanish but all digit related voices are
> taken from the english dirs... why?

Asterisk will add "sp" after the last directory name. You will need the 
following directories (or symlinks):

 /var/log/asterisk/sounds/sp
 /var/log/asterisk/sounds/phonetic/sp
 /var/log/asterisk/sounds/digit/sp
 /var/log/asterisk/sounds/letters/sp

Peter

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Polycpm SP300 problems

[Rudolf Ladyzhenskii]

Hi, all
I am trying to connect Polycom 300 to Astersik. I do not want to use FTP 
server for now, so I am tryng to set up phone manually.
Network configuration parts is OK, except that it does not ask for SIP 
server address. Any ideas where to set this?

Also i have some problems with setting up authentication.
There are settings for user name and password. How does one delete 
characters? I could not find any way to do backspace or delete!
And last question. Are user name and password on the phone should be same as 
user and secret in sip.conf file? Or those two are different things?

Thanks,
Rudolf 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Language Problems

[Anton Krall]

Guys
 
Im having a few issues with Languages.
 
Ive setup the english language is it came from default:
/var/log/asterisk/sounds 
/var/log/asterisk/sounds/phonetic
/var/log/asterisk/sounds/digit
/var/log/asterisk/sounds/letters
 
and then Spanish as
/var/log/asterisk/sounds/sp
/var/log/asterisk/sounds/sp/phonetic
/var/log/asterisk/sounds/sp/digit
/var/log/asterisk/sounds/sp/letters
 
Then one of my sip phone on sip.conf with added langauge=sp
 
Now, the normal voices ARE heard in spanish but all digit related voices are
taken from the english dirs... why?
 
her eis what the logs shows.
-- Playing 'vm-received' (language 'sp')
-- Playing 'digits/yesterday' (language 'sp')
-- Playing 'digits/at' (language 'sp')
-- Playing 'digits/11' (language 'sp')
-- Playing 'digits/20' (language 'sp')
-- Playing 'digits/4' (language 'sp')
-- Playing 'digits/p-m' (language 'sp')

Am I settings something wrong?
 
__
Anton Krall

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] FRS & *: an actual business use

[Glenn Powers]

I've noticed a growing number of stores using FRS radios. It would make 
sense to interface (via soundcard/console driver, with the nessacary 
electrical conversion) a VOX FRS radio to asterisk to allow someone in 
the office to page/talk with people on the floor or warehouse. You could 
throw that call (ie, all the radios) into a meetme conference. Then, you 
could have people in the office either dial that extension and/or have 
some of them always in that conference on a speaker phone (muted usually).

While I haven't tried this (yet!), it does seem like it would be a 
useful feature. The restriction against PSTN interconnection would be 
met UNLESS your dialplan allowed outside (PSTN) callers into that 
conference. You /could/ allow remote softphone users into the conference.

cheers,
glenn
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: FRS and GMRS via *

[Glenn Powers]

Rich Adamson wrote:
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 
CFR 95.141). There has been a lot of talk from lobbyists to clarify this 
rule, but as it stands you could conceivably connect a *private* network 
to GMRS or MURS radios (you can't make any plugins or modifications to 
an FRS radio that isn't type accepted with the radio, so connecting a 
phone line or * box would be out). The language is vague, see the 
history at http://www.provide.net/~prsg/ 
   

Would plugging into the headphone jack with a phone-patch-type device
be considered a modification for radios with vox capability?
 

I don't think that would be considered a "modification" (IANAL).
cheers,
glenn
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] playing "i" invalid context to an internal user

[Joseph]

When the call comes from outside on a certain context to play "i"
invalid extension to an external user is easy just by enclosing in an
incoming context:
exten => i,1,Answer
exten => i,2,Playback(pbx-invalid)

How to play an this context to an internal user, internal user has
access to all contexts.

In sip.conf I have
[phone number]
type=friend
context=internal
...

in extension.conf 
[internal]
include => outgoing ; include the outgoing context
include => prompts ; allow internal extens to record prompts
include => tollfree
include => iaxtel
include => invalid

[invalid]
exten => i,1,Answer
exten => i,2,Playback(pbx-invalid)
exten => i,3,Hangup()

But this doesn't work when I press any non-existent extension I get
congestion. 

-- 
#Joseph
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Transfer a call ? Am I looking fortheflashcommand ?

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Hello Jim,
> 
> thx for the answer..
> Im happy I found someone that is using flash :)

It's not perfect, but it can be useful.

> Am I right, if I transfer a call with flash, the line will be free
> afterwards ? 

Yep
 
> Would you mind to past me how you did the flash part @the
> extention file ? Also, If I use flash, do I have to setup
> anything else or just @the extention file ?

Jere's the relevant section of my dial plan:

[macro-cell_user]
exten => s,1,Playback(transfer)
exten => s,2,Flash(zap/1)
exten => s,3,SendDTMF(${ARG1})
exten => s,4,Hangup()

Good luck!

Jim.




> 
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag von Jim
> Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57
> An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for
> theflashcommand ? 
> 
> [EMAIL PROTECTED] wrote:
>> On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote:
>>> Hey Guys
>>> 
>>> Im trying to forward a call with asterisk to a regular phone.
>>> 
>>> Something like " I get a call on my regular phone, and he's trying
>>> to reach some buddy of mine.. then I tell him "wait a sec" and push
>>> "Flash" and get a other dialtone.. then I dial that other number
>>> then hangup the phone, so the one that called will be connected to
>>> where I dialed it to"... 
>>> 
>>> Some buddy of mine told me im looking for a function called "flash"
>>> 
>>> Only thing Im able to find is:
>>> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash
>>> 
>>> Im unsure how to use it now..
>>> 
>>> Let's say if I forward a call with asterisk as following: exten =>
>>> 2,1,Dial(capi/720:07812345*,18)
>>> 
>>> How would I use the flash command to transfer that call above to 078
>>> 12345* ? I have no problem transferring a call, but when Im doing
>>> this with the dial command (see above).. then my line will be busy
>> 
>> 
>> Been covered before, You can't do that on an analog line. Problem
>> comes from where you are and what flash would be working on at that
>> point. If you flash asterisk and get dialtone again, you are getting
>> the dialtone
>> from asterisk. At this point the only channel being worked is the one
>> you are on and flashing it won't help.
>> 
>> What you would need to do is get the other leg of the call to make
>> the flash.
> 
> It might be really handy to be able to specify the trunk to
> flash() as an argument. I use flash in my dialplan to
> transfer incoming calls to my cell phone when I'm out and
> about - frees up the line and reduces attenuation caused by
> an analog trombone. It'd be handy to be able to use it to
> transfer terminated calls as well.
> 
>> Of course if you where on a PRI link, you could do "hairpinning",
>> "ect" or "tromboning" and get the call taken back by the PSTN and
>> transferred to the new number.
>> --
> 
> 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Luki]

Christopher -- regarding the country checking: does your AGI also
check for mobile/cell numbers? Checking for just country codes is
trivial and I do it in the dial plan, but knowing which number is
actually a mobile call is tougher as each country does it differently.
So I'd be interested in a solution...

James -- trunking would be nice -- I have about 10 "lines" with them,
but they are residential and hence rarely do I have >2 simultaneous
calls... so I can live with that.

What is this magical 3.9 c/min about? BV officially provides "3 way
calling" which essentially means two simultaneous calls. The ATA does
the mixing, so they are indeed two independent and concurrent calls. I
do 3-day calling quite often, never had a charge for calling out on
the same "line" twice... but I do have the outgoinglimit=2 set per
line.

--Luki
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Seting up for afirst time -- can not call

[Rudolf Ladyzhenskii]

Hi, all
I am setting up Asterisk for the first time and have some problems.
Setup is very simple -- Astersik box and two Polycom SP300 phones. I will 
add bells and whistles as I go, at the moment things are very simple. No 
TFTP servers, so phones run with their default configuration.
I set up IP addresses, netmask and gateway IPs manually on the phones.

Now, I have read of problems with polycom phones. Here is my sip.conf file:
; SIP configuration file
[general]
port=5060
bindaddr=0.0.0.0
context=default
[polycom_sp300_ext101]
type=user
host=192.168.1.101
secret=101
context=default
[polycom_sp300_ext101]
type=peer
secret=101
host=192.168.1.101
context=default
callerid="Ext 101"
[polycom_sp300_ext102]
type=user
host=192.168.1.102
secret=101
context=default
[polycom_sp300_ext102]
type=peer
secret=102
host=192.168.1.102
context=default
callerid="Ext 102"
First question is about the secret. Should I set up something on teh phone? 
Is it phone password (default 456)?

Now, I am trying to have some extensions. So I have edited the 
extensions.conf file and changed the [default] section:
[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
;include => demo
exten => 101,1,Dial,(SIP/polycom_sp300_ext101)
exten => 102,1,Dial,(SIP/polycom_sp300_ext102)

The rest of the file is "as is" as it came with Asterisk.
Now I run 'reload' command as CLI.
Is ist all I have to do to be able to call between those two phones? If I 
try to call from one phone to another, after I enter first two digits '10', 
I get "connecting" on phone screen and instant busy tone.

Any help is greatly appreciated.
Thanks,
Rudolf 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Chris Ford]

I have mine set up where you just dial 6 to get out and if it is busy it 
rolls over to the next avaliable in the trunk.
- Original Message - 
From: "Greg Hill" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, February 25, 2005 7:46 PM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice


On Fri, 25 Feb 2005, James Taylor wrote:
I have two Broadvoice "lines" and there's three people in the office.
Any way to:
1) "Pool" the connections for "trunking", where any one can get a "free"
line?
2) Prevent more than 1 simultaneous call per "line"? (So I will not get
hit for 3.9 cents a minute.
have a look at ChanIsAvail, SetGroup, and CheckGroup.
Greg
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Dan Weber]

Here are the official instructions from broadvoice for setup of Asterisk. 
Other configurations are not supported.

http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Dan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk + SER

[Chris Ford]

I just installed SER last night but if you want it ot talk to Asterisk I 
found that you should install FREERADIUS Server and RADIUS CLIENT. For it to 
function properly

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

Sent: Friday, February 25, 2005 8:29 PM
Subject: [Asterisk-Users] Asterisk + SER


Hello All,
Has anyone tried Asterisk with SER.?
My main focus is billing and authentication of my endpoints.
I want Asterisk to handle all my endpoints and SER to do 
billing/accounting
stuff.

Any help will be highly appreciated.
Neel

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] open 723

[Kanishka Somaratne]

has any one implemented open 723 at http://www.readytechnology.co.uk/open/g723.1
 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk with regular analog phones

[Noah Swint]

Can regular analog phones be used and act as extensions, or does an fxs
device need to be put into place. I saw this on voip-info.org.  How
would extension setup be possible without the fxo being aware of the
name of the device?


for Analogue Phones connect to Zaptel
 channel

See Asterisk vertical service activation codes

for ZAP channels

* *Call Hold*: Normally implemented by your phone
* Unattended Transfer (or "blind transfer")
* *Consultation Hold*: Normally implemented by your phone, for
* *Unconditional Call Forwarding*
* *Attended Transfer* (or "consultative transfer"): See Asterisk
  tips zap transfer
  
* *No Answer Call Forwarding*: Implemented by yourself in the dial
  plan. See the tips & tricks page
   for ideas
* *Busy Call Forwarding*:Implemented by yourself in the dial plan.
  See the tips & tricks page
   for ideas
* *Single-Line Extension*:
* *3-way Calling*: Normally implemented by the phone
* *Incoming Call Screening*: Implemented by yourself in the dial plan
* *Find-Me*:
* *Call Pickup*: Supported in the standard installation
* *Outgoing Call Screening*: Implemented by yourself in the dial plan
* *Automatic Redial*: You should be able to implement this in the
  dial plan with some AGI support
* *Manual Redial*
* *Do-not-disturb (DND)*
* *Message waiting (MWI)*: Implemented in Asterisk, but must be
  support on the phone


-- 
http://www.FreeMiniMacs.com/?r=14335745


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] HELP NEEDED ASTERISK AND MEDIATRIX 1102

[Pedro]

Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit.  First of
all - do you have the Mediatrix Unit Manager software?  If not,
configuration will be nearly impossible.  Secondly, you will need to
configure the sip ports on the mediatrix to include "asterisk" as the
realm.  The other fields are pretty self explanatory (username,
password, etc.).  You will also want to turn off silence suppression
as it is on by default.

- Pedro


On 25 Feb 2005 20:07:04 +0100, Edward Banfa <[EMAIL PROTECTED]> wrote:
> Hello all,
> 
> Hi I would like to know how to configure a Mediatrix 1102 box to work
> with my asterisk box. I have analog phones that i would like to connect
> to my Mediatrix box and then connect the Mediatrix box to my asterisk
> box. My main problems come from the fact that I have limited experience
> with usiing the two (asterisk and the mediatrix). I know how to use
> sip.conf , but I am lost when it comes to mediatrix specific
> configuration. I have search the archives but i have not gotten any
> thing specific.
> I would really appreciate any help that can be rendered to set me in the
> right path. I am desperate here.
> Thank you all in advance
> 
> Edward
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk + SER

[Nitesh Divecha]

Hello All,

Has anyone tried Asterisk with SER.?
My main focus is billing and authentication of my endpoints.

I want Asterisk to handle all my endpoints and SER to do billing/accounting
stuff.

Any help will be highly appreciated.

Neel



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] weather asterisk@home

[Ernie Ankele]

My perl is not that great, but from your debug output, the "STREAM 
FILE" agi command never executed from the festival-weather-script.pl 
script, which should have happened right away.
Does your text2wave work? Is there a tts-.wav file in your 
var/lib/asterisk/sounds/tts dir? a tts-???.txt file?
You should be able to use this info to find out where the script is 
failing.
Ernie
On Feb 25, 2005, at 5:25 PM, James Taylor wrote:

Still no weather...
AGI Debugging Enabled
-- Executing Answer("SIP/3000-51a3", "") in new stack
-- Executing AGI("SIP/3000-51a3", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
AGI Tx >> agi_request: weather.agi
AGI Tx >> agi_channel: SIP/3000-51a3
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1109375983.318
AGI Tx >> agi_callerid: "James" <3000>
AGI Tx >> agi_dnid: 850
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: from-internal
AGI Tx >> agi_extension: 850
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-51a3", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on 
'SIP/3000-51a3'
-- Executing Macro("SIP/3000-51a3", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-51a3", "w") in new stack
-- Executing NoCDR("SIP/3000-51a3", "") in new stack
-- Executing Wait("SIP/3000-51a3", "5") in new stack
-- Executing Hangup("SIP/3000-51a3", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'SIP/3000-51a3' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
On Fri, 25 Feb 2005 16:31:43 -0700, Ernie Ankele <[EMAIL PROTECTED]> 
wrote:

Turn on debugging (agi debug) and check to see if festival is exiting 
with an error? (Maybe)
Ernie

On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
 -- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on 
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack

Any ideas?
-- James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Greg Hill]

On Fri, 25 Feb 2005, James Taylor wrote:

> I have two Broadvoice "lines" and there's three people in the office.
> Any way to:
>
> 1) "Pool" the connections for "trunking", where any one can get a "free"
> line?
> 2) Prevent more than 1 simultaneous call per "line"? (So I will not get
> hit for 3.9 cents a minute.

have a look at ChanIsAvail, SetGroup, and CheckGroup.

Greg


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] weather asterisk@home

[James Taylor]

Still no weather...
AGI Debugging Enabled
-- Executing Answer("SIP/3000-51a3", "") in new stack
-- Executing AGI("SIP/3000-51a3", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
AGI Tx >> agi_request: weather.agi
AGI Tx >> agi_channel: SIP/3000-51a3
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1109375983.318
AGI Tx >> agi_callerid: "James" <3000>
AGI Tx >> agi_dnid: 850
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: from-internal
AGI Tx >> agi_extension: 850
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-51a3", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on  
'SIP/3000-51a3'
-- Executing Macro("SIP/3000-51a3", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-51a3", "w") in new stack
-- Executing NoCDR("SIP/3000-51a3", "") in new stack
-- Executing Wait("SIP/3000-51a3", "5") in new stack
-- Executing Hangup("SIP/3000-51a3", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on  
'SIP/3000-51a3' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
On Fri, 25 Feb 2005 16:31:43 -0700, Ernie Ankele <[EMAIL PROTECTED]> wrote:

Turn on debugging (agi debug) and check to see if festival is exiting  
with an error? (Maybe)
Ernie

On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
 -- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on  
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack

Any ideas?
-- James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] weather asterisk@home

[Ernie Ankele]

Turn on debugging (agi debug) and check to see if festival is exiting 
with an error? (Maybe)
Ernie

On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
 -- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on 
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack

Any ideas?
--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: FRS radios on *

[Rich Adamson]

>  >>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
> > > CFR 95.141). There has been a lot of talk from lobbyists to clarify this
> > > rule, but as it stands you could conceivably connect a *private* network
> > > to GMRS or MURS radios (you can't make any plugins or modifications to
> > > an FRS radio that isn't type accepted with the radio, so connecting a
> > > phone line or * box would be out). The language is vague, see the
> > > history at http://www.provide.net/~prsg/
> 
>  > Would plugging into the headphone jack with a phone-patch-type device
>  > be considered a modification for radios with vox capability?
> 
> It seems dumb, but that's the way the rules are written. A patch would 
> be "other apparatus" wouldn't it?
> 
> Sec. 95.*194* (*FRS* Rule 4) *FRS* units.
> 
> (a) You may only use an FCC certified *FRS* unit. (You can identify an
> FCC certified *FRS* unit by the label placed on it by the manufacturer.)
> (b) You must not make, or have made, any internal modification to an
> *FRS* unit. Any internal modification cancels the FCC certification and
> voids your authority to operate the unit in the *FRS*.
> (c) You may not attach any antenna, power amplifier, or other
> apparatus to an *FRS* unit that has not been FCC certified as part of that
> *FRS* unit. There are no exceptions to this rule and attaching any such
> apparatus to a *FRS* unit cancels the FCC certification and voids
> everyone's authority to operate the unit in the *FRS*.
> (d) *FRS* units are prohibited from transmitting data in store-and-
> forward packet operation mode.
> 
> [61 FR 28768, June 6, 1996, as amended at 68 FR 9901, Mar. 3, 2003]

Guess if I read that literally, connecting * via a phone patch would be
acceptable since we've not modified the radio, and * voice (not voip)
is certainly not store-n-forward.

If the so-called phone patch accepted voip packets and had some sort
of modem implementation, it could probabaly send very slow speed voip
data through the headset jack. That obviously assumes the headset jack
passes sufficient audio bandwidth to even support some form of modem.

Probably not worth the effort.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem

[Mr. James W. Laferriere]

Hello Mark ,  C. & All ,  Is this device available for sale
in the US ?  All the digging I've only found outside US
mentions of sales .  Any help appreciated .  JimL
On Fri, 25 Feb 2005, Mark Elkins wrote:
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote:
Did you have to make any changes to use the premicell, or was it as simple
as an outgoing landline call?
I am looking into doing this as you can get deals where calls between chosen
numbers are free :-)
Absolutely no changes at all I did stick a Phone onto the 2-wire
input of the 'PremiCell' to check that all worked - before going via
Asterisk - but thats all.
[part of the previous message]
In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff.
Calls to Cell phones are no different to any other call...
I also added a Digium 4-port analogue card - and have a 'PremiCell'
connected to a Trunk line. The PremiCell is a fixed cell device that
gives dial-tone in the same way that a Telcom Trunk line would work -
except there is no copper to he exchange - just a stubby cellphone
antenna.  In South Africa it is MUCH MUCH cheaper to make a Cell to Cell
call than from Telcom to Cell
I'm surprised that more people do not put down a 'PremiCell' type device
and route all Cell calls out through it...
--
   +--+
   | James   W.   Laferriere | SystemTechniques | Give me VMS |
   | NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
   | [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
   +--+
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Festival - Asterisk@home

[dean collins]

Title: Festival - [EMAIL PROTECTED]








Wiley, if you follow the instructions as
listed it will work.

 

Can you post more information about what
actually isn’t working? can you post the output of your cli.

 

 

Cheers,

Dean

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Friday, February 25, 2005
3:13 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Festival
- [EMAIL PROTECTED]



 

Hello
All, 

I
installed [EMAIL PROTECTED] with no problems whatsoever.  All features so far
work great. 

However,
I have been trying to setup the festivval weather AGI script and it won't work.


I
see the script fire off in the CLI and it completes with no errors.

However,
I never hear anything on the extension. 

Does
anyone know if there is something undocumented that I should have done?


Thanks,

Wiley







___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] VoIP/Asterisk presentation

[Duane]

Ronald Hartmann wrote:
> Anyone able to get to these I am unable to get to them.

Seems I have an issue with a dead name server (the box itself) and I'll
be going in and hitting a button in the next 20 mins or so, but there's
3 other name servers so I don't know why dns doesn't just jump to
another one if one isn't reachable instead of failing like that...

-- 

Best regards,
 Duane

http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers

"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] weather asterisk@home

[James Taylor]

I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
 -- Executing Answer("SIP/3000-a844", "") in new stack
-- Executing AGI("SIP/3000-a844", "weather.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
-- AGI Script weather.agi completed, returning 0
-- Executing Hangup("SIP/3000-a844", "") in new stack
  == Spawn extension (from-internal, 850, 3) exited non-zero on  
'SIP/3000-a844'
-- Executing Macro("SIP/3000-a844", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3000-a844", "w") in new stack
-- Executing NoCDR("SIP/3000-a844", "") in new stack
-- Executing Wait("SIP/3000-a844", "5") in new stack
-- Executing Hangup("SIP/3000-a844", "") in new stack

Any ideas?
--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: FRS radios on *

[David Josephson]

Rich Adamson writes
>>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
> CFR 95.141). There has been a lot of talk from lobbyists to clarify this
> rule, but as it stands you could conceivably connect a *private* network
> to GMRS or MURS radios (you can't make any plugins or modifications to
> an FRS radio that isn't type accepted with the radio, so connecting a
> phone line or * box would be out). The language is vague, see the
> history at http://www.provide.net/~prsg/
> Would plugging into the headphone jack with a phone-patch-type device
> be considered a modification for radios with vox capability?
It seems dumb, but that's the way the rules are written. A patch would 
be "other apparatus" wouldn't it?

Sec. 95.*194* (*FRS* Rule 4) *FRS* units.
(a) You may only use an FCC certified *FRS* unit. (You can identify an
FCC certified *FRS* unit by the label placed on it by the manufacturer.)
(b) You must not make, or have made, any internal modification to an
*FRS* unit. Any internal modification cancels the FCC certification and
voids your authority to operate the unit in the *FRS*.
(c) You may not attach any antenna, power amplifier, or other
apparatus to an *FRS* unit that has not been FCC certified as part of that
*FRS* unit. There are no exceptions to this rule and attaching any such
apparatus to a *FRS* unit cancels the FCC certification and voids
everyone's authority to operate the unit in the *FRS*.
(d) *FRS* units are prohibited from transmitting data in store-and-
forward packet operation mode.
[61 FR 28768, June 6, 1996, as amended at 68 FR 9901, Mar. 3, 2003]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] VM+Realtime config

[Time Bandit]

> 1) when Asterisk try to build the mailbox directory under the path :
> /var/spool/asterisk/... 
Don't know about realtime, but in "standard" version, the directory is
built the first time you leave a message to this mailbox

hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SER vs. Asterisk - call in progress to PSTN

[Mik Cheez]

We're having a problem with Asterisk when we try to pass a call off to a 
Lucent PSTN using SIP.  This behavior does not exist with SER:

With Asterisk
An ISDN call is started, at the T1 level we receive “call proceeding” 
and immediately we receive a “Call in Progress” just like the far end 
party has answered.

With SER
An ISDN call is started, at the T1 level we receive “call proceeding” 
which is OK, and when the call fails we get the cause code of why the 
call failed

Any suggestions as to how Asterisk can wait for the PSTN to receive a 
call in progress?

Best regards
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[James Taylor]

I have two Broadvoice "lines" and there's three people in the office.
Any way to:
1) "Pool" the connections for "trunking", where any one can get a "free"  
line?
2) Prevent more than 1 simultaneous call per "line"? (So I will not get  
hit for 3.9 cents a minute.

I'd like to use the country code AGI.
James
On Fri, 25 Feb 2005 15:52:28 -0500, Christopher McBee <[EMAIL PROTECTED]>  
wrote:

Here is a copy of my config that works great with broadvoice.  I also
have an AGI that I wrote to verify country codes so your users can't
call countries that aren't included in broadvoices plan.  If you want
that too, just let me know.
Sip.conf
-
; Inbound broadvoice calls
register => 8029041486:[EMAIL PROTECTED]/8029041486
[Broadvoice]
type=friend
username=8029041486
fromuser=8029041486
secret=zjfg9f18fh
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
port=5060
dtmfmode=inband
insecure=very
permit=147.135.0.128/32
qualify=yes
canreinvite=yes
nat=no

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, February 25, 2005 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls on this trunk
work but NOT both.  The "sip show registry" command shows everything as
it should be.
The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no
My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
If I remove type=peer from [Broadvoice] in sip.conf incoming calls work
great but outgoing calls don't work.  If i leave type=peer in there,
outgoing calls work great but incoming calls get routed to Broadvoice's
Voicemail . . .
Roger Hanson wrote:
- Original Message - From: <[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice

I have configured asterisk with the AMP php configuration utility.  I

am able to make outgoing calls through broadvoice but incoming calls
are sent to BV's Voicemail and never actually enter the IVR.  When I
show sip debug info through the asterisk prompt it actually reads the

incoming call from BV but then issues a busy signal sending the call
to BV's voicemail.
I also modified extensions.conf as follows:
[from-sip-external]
include => from-pstn
I have set up my sip trunk in AMP as follows:
Trunk Name: Broadvoice
Peer Details:
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=21
host=sip.broadvoice.com
qualify=yes
secret=password
type=peer
username=21
My Incoming Settings are:
User Context: sip.broadvoice.com
User Details:
context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
nat=yes
secret=password
user=21
username=21
My register string:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

Something to double check and something to try (in that order):
1.  check your password.  It's not the password you registered at
their website with.  They send you an email with a different password
in it you need to use.  The password you registered at their website
is just for logging into their website.
2.  Try using a standard registration string - not the one they show
you.  Use number:[EMAIL PROTECTED] instead of the one they
show you on the website.
See if one of those things is the trouble.
If that doesn't work, look at "sip show registry" and see what's
registered.
asterisk*CLI> sip show registry
Host  UsernameRefresh
State
sip.broadvoice.com:5060 952225  15 Registered
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] "click to dial extension number" functionality ?

[dean collins]

http://www.voip-info.org/wiki-Microappliances+SIP+Active-X+Client

 

http://www.microappliances.com/site/html/index.php?section=Products&page=clienthowto.php

 

I haven’t tried it though, let me
know how it goes.

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terje Myhre
Sent: Friday, February 25, 2005
6:10 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users]
"click to dial extension number" functionality ? 



 

 

Hello, 

 

We would like to : 

 

By any web-user (ms explorer) to be able to call from a
web-page to a certain number/extension connected to one specific asterisk. 

 

Almost as a web-based “auto-attendant”
functionality.  

 

Hence: 


 surf to the specific web-site
 enter the extension digits in a
 web-interface
 get connected – with in-
 and out-sound through the web-browser 


 

Do anyone know what would be the simplest / best way to
implement this functionality ? 

 

Br, 


Terje Myhre 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CallerID Name and Digium TE405P

[B. J. Bomar]

Hello all, I am 
looking at replacing our current Cisco PRI gateway with a new server with a 
TE405P card.  My primary concern is receiving CallerID Name info on 
the D-Channel.  Does anyone have any experience terminating a local Qwest 
PRI from a 5ES switch into the TE405P or similar?  We are also looking at 
getting a McLeod PRI, which I believe is off of a DMS100 switch.  Any 
thoughts would be greatly appreciated.
 
B. 
J.
 
 
 
 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] VM+Realtime config

[mohammad]

Hi ALL;
 
When I insert data to Vm table in Realtime config, 
I canot see any directory built under : /var/spool/asterisk
 
1) when Asterisk try to build the mailbox 
directory under the path : /var/spool/asterisk/...
 
2) Where is source code that tells Asterisk to 
build that directory under: /var/spool/asterisk/...
 
 
 
Thanks
Mohammad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Directory config...

[Francois Meehan]

Right on!

Have a good week-end!

Francois

>> How do I config Asterisk so when the directory cmd is used, the name of
>> the found entry comes from a pre-record gsm file instead of being
>> spelled
>> letter by letter?
> If the user as recorded is name, this file will be used. When it's not
> recorded, * will spell it.
>
> Dial to your voicemail and navigate thru the menu to record your name.
>
> hth
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Spam detection software, running on the system "zeus.avanzada7.com", has
> identified this incoming email as possible spam.  The original message
> has been attached to this so you can view it (if it isn't spam) or label
> similar future email.  If you have any questions, see
> the administrator of that system for details.
>
> Content preview:  > How do I config Asterisk so when the directory cmd
>   is used, the name of > the found entry comes from a pre-record gsm
>   file instead of being spelled > letter by letter? If the user as
>   recorded is name, this file will be used. When it's not recorded, *
>   will spell it. [...]
>
> Content analysis details:   (0.6 points, 5.0 required)
>
>  pts rule name  description
>  --
> --
>  0.5 FROM_ENDS_IN_NUMS  From: ends in numbers
>  0.0 RCVD_BY_IP Received by mail server with no name
>  0.1 FORGED_RCVD_HELO   Received: contains a forged HELO
>
>
>
>



Random Thought:
---
I just remembered something about a TOAD!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does the g.729 registration program work?

[Mik Cheez]

Thanks for the clarification.  In that case the following should only be 
considered for development.

Steven Critchfield wrote:
On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote:
 

There is an open source version of the license:

You can view the licensing information at the following:

more details can be found on http://www.voip-info.org
   

http://www.intel.com/software/products/ipp/speech_code_lin_eula.htm
And in section 3.E of the license above listed, you still must obtain
licenses from the patent holders. 

Also, arguably, section 8. removes the ability to use in most phone
systems. 

While it MAY be highly unlikely that you get prosecuted for using the
code on your home machine. But if you start to scale up to any
significant port usage, you may find yourself battling to not lose the
company on a simple licensable part.
Not to mention the fact that you will possibly get flamed for not having
gone the legal and community building route of buying from Digium.
 

Steven Critchfield wrote:
   

On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote:
 

I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:
- There's a config file, though I've seen no mention of it
- The actual binary shared library is modified
- The system contacts Digium every time you start asterisk
In the last case nothing is changed at all and I'm fine.
  

   

It is based on a machine unique key created by querying your hardware.
You will not be able to share your licenses between machines. You will
need to buy licenses for each machine you deploy on.
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk in front of Toshiba CTX

[Daniel Burget]

I have googled, and wiki’ed until blue. Is it possible
to put T1---*Toshiba CTX ? I have a TE405P, with one interface programmed
for the T1, I am not sure how to program the 2nd port to mimick the
T1 to the Toshiba. The Zapata.conf 

 

[channels]

switchtype=national

context=from-pstn

signalling=pri_cpe

usecallerid=asreceived

echocancel=yes

echocancelwhenbridged=no

echotraining=400

overlapdial=yes

immediate=no

group=0

channel => 1-23

 

Zaptel.conf

 

bchan=1-23

dchan=24

span=1,0,0,esf,b8zs

 

This works for the T1 into *, or Into Toshiba. I want the
calls to go into *, if they don’t match exten.conf to go to the Toshiba.
If the Toshiba dials out, It goes into * and out via sip. So, the T1 will be
for internal DID only.

 

Is this possible, or am I chasing a dream?

 

Thanks much!

 

Dan






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RE:Avaya Partner ACS3 and Asterisk

[Jason Kawakami]

-Original Message-
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function and be able to dial a number like that.  I can get all
the asterisk stuff set up to do this, i just need to know if anyone knows
what setting to use in the partner system to make it bring up a dialtone
so i can dial.

-you will have to dial an extension number on the avaya that goes to the fxo
in *.  If * is set up to do so, you should be able to just dial the number
that you wish to be routed to the voip provider and then the dialplan in *
will take care of it.

Imagine that you dial extension 10 on the partner.

* answers the call
Reads the DTMF from the caller (prompt optional)
Sets the read DTMF as the dial string 
Issues a dial to the voip provider

Should work I think.

Jason Kawakami
www.optellabs.com   
Salt Lake City, UT



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[[EMAIL PROTECTED]]

Apparently the combination of the correct registry string and 
insecure=very fixed it.  Just as you said.

Thanks!
Roger Hanson wrote:
- Original Message - From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, February 25, 2005 9:20 AM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice


Great!  It works now!!  Thanks so much.
What was it that made it work?  Share the knowledge with the world.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Roger Hanson]

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

Sent: Friday, February 25, 2005 9:20 AM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice


Great!  It works now!!  Thanks so much.
What was it that made it work?  Share the knowledge with the world. 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SetCIDNum using SIP?

[Michael Bielicki]

He was talking about SIP


On Fri, 25 Feb 2005 16:09:27 -0500, Kevin Collins <[EMAIL PROTECTED]> wrote:
> Nabeel,
> 
> Works for me see below.
> 
> exten => _1NXXNXX,1,SetCIDNum(55,a)
> exten => _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt)
> 
> Kevin
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel 
> Jafferali
> Sent: Friday, February 25, 2005 3:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] SetCIDNum using SIP?
> 
> I am experimenting with my * server to use SIP with my long-distance
> providers instead of IAX, so that the media path is from the end user
> straight to the provider's gateway (hopefully reducing my bandwidth
> consumption). I have it working with VoicePulse Connect but SetCIDNum
> doesn't appear to work. Is this something with VoicePulse Connect only
> or is it generally difficult to set the CallerID info on a per-call
> basis when using SIP?
> 
> Nabeel
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 


-- 
Michal Bielicki
http://www.asterisk.com.pl/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[[EMAIL PROTECTED]]

Great!  It works now!!  Thanks so much.
Roger Hanson wrote:
- Original Message - From: "Robert Webb" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
; <[EMAIL PROTECTED]>
Sent: Friday, February 25, 2005 2:49 PM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice


On Fri, 25 Feb 2005 14:42:09 +
 "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls on this 
trunk work but NOT both.  The "sip show registry" command shows 
everything as it should be.

The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no
My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
If I remove type=peer from [Broadvoice] in sip.conf incoming calls 
work great but outgoing calls don't work. If i leave type=peer in 
there, outgoing calls work great but incoming calls get routed to 
Broadvoice's Voicemail . . .

Roger Hanson wrote:
- Original Message - From: <[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice

I have configured asterisk with the AMP php configuration utility. 
I am able to make outgoing calls through broadvoice but incoming 
calls are sent to BV's Voicemail and never actually enter the IVR. 
When I show sip debug info through the asterisk prompt it actually 
reads the incoming call from BV but then issues a busy signal 
sending the call to BV's voicemail.

I also modified extensions.conf as follows:
[from-sip-external]
include => from-pstn
I have set up my sip trunk in AMP as follows:
Trunk Name: Broadvoice
Peer Details:
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=21
host=sip.broadvoice.com
qualify=yes
secret=password
type=peer
username=21
My Incoming Settings are:
User Context: sip.broadvoice.com
User Details:
context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
nat=yes
secret=password
user=21
username=21
My register string:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

Something to double check and something to try (in that order):
1.  check your password.  It's not the password you registered at 
their website with.  They send you an email with a different 
password in it you need to use.  The password you registered at 
their website is just for logging into their website.

2.  Try using a standard registration string - not the one they show 
you.  Use number:[EMAIL PROTECTED] instead of the one they 
show you on the website.

See if one of those things is the trouble.
If that doesn't work, look at "sip show registry" and see what's 
registered.
asterisk*CLI> sip show registry
Host  Username Refresh State
sip.broadvoice.com:5060 952225  15 Registered

Mine ONLY works both directions when I use a normal registration 
string. And remember, don't use the password you signed up with on 
their website.  They email you a different password you need to use in 
your Asterisk configurations.

I know some people have to use the funky registration string, but it 
wouldn't work for me (and some others).  Also, I know of some others 
that couldn't get it to work without the line:  insecure=very

Here's my sip:
register=myphonenumber:[EMAIL PROTECTED]

[myphonenumber]
type=friend
secret=mypassword
regexten=myphonenumber
insecure=very
host=sip.broadvoice.com
fromuser=myphonenumber
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SetCIDNum using SIP? Ignore my last post

[Kevin Collins]

Nabeel,

Ignore my last post. Missed the SIP part of your question.
 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Collins
Sent: Friday, February 25, 2005 4:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SetCIDNum using SIP?

Nabeel,

Works for me see below.

exten => _1NXXNXX,1,SetCIDNum(55,a)
exten => _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt) 

Kevin

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali
Sent: Friday, February 25, 2005 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SetCIDNum using SIP?

I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to work. Is this something with VoicePulse Connect only
or is it generally difficult to set the CallerID info on a per-call
basis when using SIP?

Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


smime.p7s
Description: S/MIME cryptographic signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Roger Hanson]

- Original Message - 
From: "Robert Webb" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
; <[EMAIL PROTECTED]>
Sent: Friday, February 25, 2005 2:49 PM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice


On Fri, 25 Feb 2005 14:42:09 +
 "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls on this trunk 
work but NOT both.  The "sip show registry" command shows everything 
as it should be.

The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no
My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
If I remove type=peer from [Broadvoice] in sip.conf incoming calls 
work great but outgoing calls don't work. If i leave type=peer in 
there, outgoing calls work great but incoming calls get routed to 
Broadvoice's Voicemail . . .

Roger Hanson wrote:
- Original Message - From: <[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice

I have configured asterisk with the AMP php configuration utility. 
I am able to make outgoing calls through broadvoice but incoming 
calls are sent to BV's Voicemail and never actually enter the IVR. 
When I show sip debug info through the asterisk prompt it actually 
reads the incoming call from BV but then issues a busy signal 
sending the call to BV's voicemail.

I also modified extensions.conf as follows:
[from-sip-external]
include => from-pstn
I have set up my sip trunk in AMP as follows:
Trunk Name: Broadvoice
Peer Details:
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=21
host=sip.broadvoice.com
qualify=yes
secret=password
type=peer
username=21
My Incoming Settings are:
User Context: sip.broadvoice.com
User Details:
context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
nat=yes
secret=password
user=21
username=21
My register string:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

Something to double check and something to try (in that order):
1.  check your password.  It's not the password you registered at 
their website with.  They send you an email with a different password 
in it you need to use.  The password you registered at their website 
is just for logging into their website.

2.  Try using a standard registration string - not the one they show 
you.  Use number:[EMAIL PROTECTED] instead of the one they 
show you on the website.

See if one of those things is the trouble.
If that doesn't work, look at "sip show registry" and see what's 
registered.
asterisk*CLI> sip show registry
Host  Username Refresh State
sip.broadvoice.com:5060 952225  15 Registered

Mine ONLY works both directions when I use a normal registration string. 
And remember, don't use the password you signed up with on their 
website.  They email you a different password you need to use in your 
Asterisk configurations.

I know some people have to use the funky registration string, but it 
wouldn't work for me (and some others).  Also, I know of some others 
that couldn't get it to work without the line:  insecure=very

Here's my sip:
register=myphonenumber:[EMAIL PROTECTED]

[myphonenumber]
type=friend
secret=mypassword
regexten=myphonenumber
insecure=very
host=sip.broadvoice.com
fromuser=myphonenumber
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SetCIDNum using SIP?

[Kevin Collins]

Nabeel,

Works for me see below.

exten => _1NXXNXX,1,SetCIDNum(55,a)
exten => _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt) 

Kevin

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali
Sent: Friday, February 25, 2005 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SetCIDNum using SIP?

I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to work. Is this something with VoicePulse Connect only
or is it generally difficult to set the CallerID info on a per-call
basis when using SIP?

Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


smime.p7s
Description: S/MIME cryptographic signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Fedora Core 3?

[Glenn Powers]

Race Vanderdecken wrote:
I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6 I could understand.)
 

http://www.linuxprinting.org/pipermail/okidata-list/2004q2/000359.html
cheers,
glenn
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[Robert Webb]

On Fri, 25 Feb 2005 14:42:09 +
 "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls 
on this trunk work but NOT both.  The "sip show registry" 
command shows everything as it should be.

The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no
My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
If I remove type=peer from [Broadvoice] in sip.conf 
incoming calls work great but outgoing calls don't work. 
If i leave type=peer in there, outgoing calls work great 
but incoming calls get routed to Broadvoice's Voicemail . 
. .

Roger Hanson wrote:
- Original Message - From: 
<[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice


I have configured asterisk with the AMP php 
configuration utility.  I 
am able to make outgoing calls through broadvoice but 
incoming calls 
are sent to BV's Voicemail and never actually enter the 
IVR.  When I 
show sip debug info through the asterisk prompt it 
actually reads the 
incoming call from BV but then issues a busy signal 
sending the call 
to BV's voicemail.

I also modified extensions.conf as follows:
[from-sip-external]
include => from-pstn
I have set up my sip trunk in AMP as follows:
Trunk Name: Broadvoice
Peer Details:
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=21
host=sip.broadvoice.com
qualify=yes
secret=password
type=peer
username=21
My Incoming Settings are:
User Context: sip.broadvoice.com
User Details:
context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
nat=yes
secret=password
user=21
username=21
My register string:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

Something to double check and something to try (in that 
order):

1.  check your password.  It's not the password you 
registered at 
their website with.  They send you an email with a 
different password 
in it you need to use.  The password you registered at 
their website 
is just for logging into their website.

2.  Try using a standard registration string - not the 
one they show 
you.  Use number:[EMAIL PROTECTED] instead of 
the one they 
show you on the website.

See if one of those things is the trouble.
If that doesn't work, look at "sip show registry" and 
see what's 
registered.
asterisk*CLI> sip show registry
Host  Username 
  Refresh 
State
sip.broadvoice.com:5060 952225  15 
Registered

Did you try changing what Roger wrote in number two above?
2. Try using a standard registration string - not the one 
they show you. Use number:[EMAIL PROTECTED] 
instead of the one they show you on the website.

I see you did not change that in your second post.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk With Broadvoice

[Christopher McBee]

Here is a copy of my config that works great with broadvoice.  I also
have an AGI that I wrote to verify country codes so your users can't
call countries that aren't included in broadvoices plan.  If you want
that too, just let me know.


Sip.conf
-
; Inbound broadvoice calls
register => 8029041486:[EMAIL PROTECTED]/8029041486


[Broadvoice]
type=friend
username=8029041486
fromuser=8029041486
secret=zjfg9f18fh
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
port=5060
dtmfmode=inband
insecure=very
permit=147.135.0.128/32
qualify=yes
canreinvite=yes
nat=no


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, February 25, 2005 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice

OK,

After checking into this, I have found the following:

I can set it up so either incoming or outgoing sip calls on this trunk 
work but NOT both.  The "sip show registry" command shows everything as 
it should be.

The section from my sip.conf is as follows:

[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no

My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

If I remove type=peer from [Broadvoice] in sip.conf incoming calls work 
great but outgoing calls don't work.  If i leave type=peer in there, 
outgoing calls work great but incoming calls get routed to Broadvoice's 
Voicemail . . .


Roger Hanson wrote:

>
> - Original Message - From: <[EMAIL PROTECTED]>
> To: 
> Sent: Thursday, February 24, 2005 10:12 PM
> Subject: [Asterisk-Users] Asterisk With Broadvoice
>
>
>> I have configured asterisk with the AMP php configuration utility.  I

>> am able to make outgoing calls through broadvoice but incoming calls 
>> are sent to BV's Voicemail and never actually enter the IVR.  When I 
>> show sip debug info through the asterisk prompt it actually reads the

>> incoming call from BV but then issues a busy signal sending the call 
>> to BV's voicemail.
>>
>> I also modified extensions.conf as follows:
>> [from-sip-external]
>> include => from-pstn
>>
>> I have set up my sip trunk in AMP as follows:
>>
>> Trunk Name: Broadvoice
>> Peer Details:
>> dtmfmode=inband
>> fromdomain=sip.broadvoice.com
>> fromuser=21
>> host=sip.broadvoice.com
>> qualify=yes
>> secret=password
>> type=peer
>> username=21
>>
>> My Incoming Settings are:
>> User Context: sip.broadvoice.com
>> User Details:
>> context=from-pstn
>> dtmfmode=inband
>> fromdomain=sip.broadvoice.com
>> host=sip.broadvoice.com
>> nat=yes
>> secret=password
>> user=21
>> username=21
>>
>> My register string:
>> [EMAIL PROTECTED]:[EMAIL PROTECTED]
>>
>>
>
> Something to double check and something to try (in that order):
>
> 1.  check your password.  It's not the password you registered at 
> their website with.  They send you an email with a different password 
> in it you need to use.  The password you registered at their website 
> is just for logging into their website.
>
> 2.  Try using a standard registration string - not the one they show 
> you.  Use number:[EMAIL PROTECTED] instead of the one they 
> show you on the website.
>
> See if one of those things is the trouble.
>
> If that doesn't work, look at "sip show registry" and see what's 
> registered.
> asterisk*CLI> sip show registry
> Host  UsernameRefresh 
> State
> sip.broadvoice.com:5060 952225  15 Registered
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 2/22/2005
 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SetCIDNum using SIP?

[Nabeel Jafferali]

I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to work. Is this something with VoicePulse Connect only
or is it generally difficult to set the CallerID info on a per-call
basis when using SIP?

Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk With Broadvoice

[[EMAIL PROTECTED]]

OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls on this trunk 
work but NOT both.  The "sip show registry" command shows everything as 
it should be.

The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no
My registry string is:
[EMAIL PROTECTED]:[EMAIL PROTECTED]
If I remove type=peer from [Broadvoice] in sip.conf incoming calls work 
great but outgoing calls don't work.  If i leave type=peer in there, 
outgoing calls work great but incoming calls get routed to Broadvoice's 
Voicemail . . .

Roger Hanson wrote:
- Original Message - From: <[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice

I have configured asterisk with the AMP php configuration utility.  I 
am able to make outgoing calls through broadvoice but incoming calls 
are sent to BV's Voicemail and never actually enter the IVR.  When I 
show sip debug info through the asterisk prompt it actually reads the 
incoming call from BV but then issues a busy signal sending the call 
to BV's voicemail.

I also modified extensions.conf as follows:
[from-sip-external]
include => from-pstn
I have set up my sip trunk in AMP as follows:
Trunk Name: Broadvoice
Peer Details:
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=21
host=sip.broadvoice.com
qualify=yes
secret=password
type=peer
username=21
My Incoming Settings are:
User Context: sip.broadvoice.com
User Details:
context=from-pstn
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
nat=yes
secret=password
user=21
username=21
My register string:
[EMAIL PROTECTED]:[EMAIL PROTECTED]

Something to double check and something to try (in that order):
1.  check your password.  It's not the password you registered at 
their website with.  They send you an email with a different password 
in it you need to use.  The password you registered at their website 
is just for logging into their website.

2.  Try using a standard registration string - not the one they show 
you.  Use number:[EMAIL PROTECTED] instead of the one they 
show you on the website.

See if one of those things is the trouble.
If that doesn't work, look at "sip show registry" and see what's 
registered.
asterisk*CLI> sip show registry
Host  UsernameRefresh 
State
sip.broadvoice.com:5060 952225  15 Registered

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] E100P to Valiant E1-PRI GSM gateway

[Rusty Shackleford]

Looking for zaptel/zapata configuration parameters to successfully
communicate with a Valiant GSM gateway as above.
Surely someone has done this?

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 02/22/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Transposed ringing

[Robert Webb]

On Fri, 25 Feb 2005 12:09:16 -0800
 Trevor Peirce <[EMAIL PROTECTED]> wrote:
I don't suppose anyone might know why I hear ringing 
transposed over itself when I place a call out via PRI?

SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much because it's 
lined up right, but other times you'll hear a really long 
ring (starts sounding normal, then sounds "weird" -- like 
two rings played at the same time, then sound normal) or 
quick choppy rings.  It seems very much like there is a 
ring from the PRI and another ring from asterisk or the 
Sipura phones & ATAs.

Any clues would be appreciated.
Trevor
It is exactly what you are stating. You are getting the 
rig tone generated by the local device and the normal ring 
genterated by the local telco. I have this exact same 
problem with an IAX softphone. I just turned off the local 
ring.

Sorry, but I can be no more help than that at this point.
Robert
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Festival - Asterisk@home

[Wiley Siler]

Title: Festival - [EMAIL PROTECTED]



figured it out.  Stopped using the example 
festival-weather.script.pl and used the festival-script.pl that is in the 
directory already.
 
Works good.  Is the voice customizable?  Does 
memory on the box play a part in quality?
 
Thanks,
Wiely
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan 
HobbsSent: Friday, February 25, 2005 1:19 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Festival - [EMAIL PROTECTED]

Does your Festival installation work ok?  (run the 
tests/example scripts that came with the installation).  I installed 
Festival, and according to the installation scripts all went well, however none 
of the tests/example routines would work - I kept getting 'Segmentation Faults' 
whenever I ran them.
 
I gave up on Festival (unable to resolve the segmentation 
faults in a reasonable number of hours of effort), but if you manage to get it 
running let me know how!
 
Jonathan
 
 

  - Original Message - 
  From: 
  Wiley 
  Siler 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: February 25, 2005 3:13 PM
  Subject: [Asterisk-Users] Festival - 
  [EMAIL PROTECTED]
  
  Hello All, 
  I installed [EMAIL PROTECTED] with no problems 
  whatsoever.  All features so far work great. 
  However, I have been trying to setup the festivval 
  weather AGI script and it won't work. 
  I see the script fire off in the CLI and it 
  completes with no errors. However, I never 
  hear anything on the extension. 
  Does anyone know if there is something undocumented 
  that I should have done? 
  Thanks, Wiley 
  
  

  ___Asterisk-Users 
  mailing 
  listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:   
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] WebVMail Woirks but No Audio

[Richard J. Sears]

Hi Assaf - 

Already did that  - the audio app location shows as a broken link on the
page and plays nothing.


On Fri, 25 Feb 2005 14:06:41 -0500
"Assaf Benharoosh" <[EMAIL PROTECTED]> wrote:

>  I had this issue- it's security on the files. I put a cron job that do 
> /bin/chmod 777 /var/spool/asterisk/voicemail/default -R 
> evey 1 minute, but there may be a cleaner solution.
> 
> 
> Assaf Benharoosh
> MCP, MCSA, MCSE
> [EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Richard J.
> Sears
> Sent: Friday, February 25, 2005 11:20 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] WebVMail Woirks but No Audio
> 
> Hi Everyone - 
> 
> I have webvmail up and running, I can see the messages, forward them,
> pretty much everything but listen to them.
> 
> Here is what I see in my logs:
> 
> 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET
> /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default
> &password=12&msgid=&format=gsm&dontcasheme=4624.gsm HTTP/1.1"
> 200 9438 "-" "contype"
> 
> 
> But the box at the bottom shows up as a broken link.
> 
> Any ideas...?
> 
> 
> Thanks
> 
> 
> **
> Richard J. Sears
> Vice President 
> American Internet Services  
> 
> [EMAIL PROTECTED]
> http://www.adnc.com
> 
> 858.576.4272 - Phone
> 858.427.2401 - Fax
> INOC-DBA - 6130
> 
> 
> I fly because it releases my mind
> from the tyranny of petty things . . 
> 
> 
> "Work like you don't need the money, love like you've
> never been hurt and dance like you do when nobody's
> watching."
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


**
Richard J. Sears
Vice President 
American Internet Services  

[EMAIL PROTECTED]
http://www.adnc.com

858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130


I fly because it releases my mind 
from the tyranny of petty things . . 


"Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching."

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 304

[David Josephson]

Daniel Nystrom wrote
It seems like the Radio discussions is closing in on something I was
interested in.
How about controlling 30 2-way radios via E1 and 30-channel "Mux"
(channel bank?) with E&M signalling?
I think the Mux uses CAS and each channel has Audio out, PTT, Audio
IN, Busy. 6-wire connection i guess?
That should be a really nice setup with Asterisk!
Anyone tried something like this?
We didn't do it with Asterisk, but a group I have been part of for about 30 
years is doing this in northern California using homemade crossbar switches and 
Mitel PBXs and T1. Obviously we are more than a little interested in Asterisk.
One last time - go read Jim Dixon's app_rpt stuff, that looks like the place to 
start.
Now we need someone to write a module to emulate an IMTS base station and our 
telco will be complete ;)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Transposed ringing

[Trevor Peirce]

I don't suppose anyone might know why I hear ringing transposed over 
itself when I place a call out via PRI?

SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much because it's lined up right, 
but other times you'll hear a really long ring (starts sounding normal, 
then sounds "weird" -- like two rings played at the same time, then 
sound normal) or quick choppy rings.  It seems very much like there is a 
ring from the PRI and another ring from asterisk or the Sipura phones & 
ATAs.

Any clues would be appreciated.
Trevor
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Festival - Asterisk@home

[Jonathan Hobbs]

Title: Festival - [EMAIL PROTECTED]



Does your Festival installation work ok?  (run the 
tests/example scripts that came with the installation).  I installed 
Festival, and according to the installation scripts all went well, however none 
of the tests/example routines would work - I kept getting 'Segmentation Faults' 
whenever I ran them.
 
I gave up on Festival (unable to resolve the segmentation 
faults in a reasonable number of hours of effort), but if you manage to get it 
running let me know how!
 
Jonathan
 
 

  - Original Message - 
  From: 
  Wiley 
  Siler 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: February 25, 2005 3:13 PM
  Subject: [Asterisk-Users] Festival - 
  [EMAIL PROTECTED]
  
  Hello All, 
  I installed [EMAIL PROTECTED] with no problems 
  whatsoever.  All features so far work great. 
  However, I have been trying to setup the festivval 
  weather AGI script and it won't work. 
  I see the script fire off in the CLI and it 
  completes with no errors. However, I never 
  hear anything on the extension. 
  Does anyone know if there is something undocumented 
  that I should have done? 
  Thanks, Wiley 
  
  

  ___Asterisk-Users 
  mailing 
  listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:   
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does the g.729 registration program work?

[Martijn van Oosterhout]

On Fri, Feb 25, 2005 at 11:24:21AM -0600, Steven Critchfield wrote:
> It is based on a machine unique key created by querying your hardware.
> You will not be able to share your licenses between machines. You will
> need to buy licenses for each machine you deploy on.

You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from the previous image to keep the licences. Obviously reformatting
the disk and reregistering is not going to work.

So what will???

Thanks in advance,
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Festival - Asterisk@home

[Wiley Siler]

Title: Festival - [EMAIL PROTECTED]






Hello All,


I installed [EMAIL PROTECTED] with no problems whatsoever.  All features so far work great.


However, I have been trying to setup the festivval weather AGI script and it won't work.


I see the script fire off in the CLI and it completes with no errors.

However, I never hear anything on the extension.


Does anyone know if there is something undocumented that I should have done?


Thanks,

Wiley



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Video Support Not Working

[Nathan Martinez]

Title: Video Support Not Working






Hello,


I have a couple of video phones that I am trying to get setup.  I have used these phones with sipphone.com and they work great.  Now I am trying to get them to work with my * server and I am having problems.  The voice portion seems to work fine, but I can not get video to work.  These phones and my * server are all on the same subnet connected to the same Ethernet switch.  I have included my sip.conf below for reference.  I also get a message in the * console each time I establish a call:  WARNING[9298]: chan_sip.c:6134 receive_info: Unable to parse INFO message from [EMAIL PROTECTED] Content.

Any help that anyone can provide is very much appreciated.


Thank you,

Nathan



[general]

context=sip

port=5060

bindaddr=0.0.0.0

srvlookup=no


disallow=all

allow=g729

allow=h263

allow=h261

allow=alaw

allow=ulaw

allow=iLBC


musicclass=default

language=en

rtptimeout=60

rtpholdtimeout=300

useragent=Asterisk PBX

promiscredir = yes

videosupport=yes


[101]

type=friend

context=sip

username=101

secret=101

fromuser=101

callerid="101" <101>

host=dynamic

nat=no

canreinvite=yes

qualify=200

dtmfmode=rfc2833


[201]

type=friend

context=sip

username=201

secret=201

fromuser=201

callerid="201" <201>

host=dynamic

nat=no

canreinvite=yes

qualify=200

dtmfmode=rfc2833




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Wheres the Math application

[John Voss]

From the docs on this command it should be available in 1.0.1. 

I don't find it under the apps directory even after doing an update -d (which I 
understand will add missing files or diretories)

I have also downloaded ver 1.0.5 and looked in its apps directory. It isn't 
there either.

Any suggestions?
-- 
___
Sign-up for Ads Free at Mail.com
http://promo.mail.com/adsfreejump.htm

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] WebVMail Woirks but No Audio

[Assaf Benharoosh]

 I had this issue- it's security on the files. I put a cron job that do 
/bin/chmod 777 /var/spool/asterisk/voicemail/default -R 
evey 1 minute, but there may be a cleaner solution.


Assaf Benharoosh
MCP, MCSA, MCSE
[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard J.
Sears
Sent: Friday, February 25, 2005 11:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] WebVMail Woirks but No Audio

Hi Everyone - 

I have webvmail up and running, I can see the messages, forward them,
pretty much everything but listen to them.

Here is what I see in my logs:

192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET
/vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default
&password=12&msgid=&format=gsm&dontcasheme=4624.gsm HTTP/1.1"
200 9438 "-" "contype"


But the box at the bottom shows up as a broken link.

Any ideas...?


Thanks


**
Richard J. Sears
Vice President 
American Internet Services  

[EMAIL PROTECTED]
http://www.adnc.com

858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130


I fly because it releases my mind
from the tyranny of petty things . . 


"Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching."

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] HELP NEEDED ASTERISK AND MEDIATRIX 1102

[Edward Banfa]

Hello all,

Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix box to my asterisk
box. My main problems come from the fact that I have limited experience
with usiing the two (asterisk and the mediatrix). I know how to use
sip.conf , but I am lost when it comes to mediatrix specific
configuration. I have search the archives but i have not gotten any
thing specific.
I would really appreciate any help that can be rendered to set me in the
right path. I am desperate here.
Thank you all in advance

Edward

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FW: Getting PHP Config to work?

[Tzafrir Cohen]

On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote:
> Hi,
> 
> Thanks for the batchfile type, it's a handy one.
> 
> I'm not editing over the internet, just local LAN for testing ATM. Protected
> via firewall.
> 
> Would it not be fairly secure using a combination of the following:
> .htaccess file
> VPN?
> https access?
> Limit apache to only allow certain IP's?
> And the public keys thing.

Secure agains what? What are the threats you consider?

VPN and/or limit of IP addresses (in iptables or in apache's config)
would serve to allow access only from certain addresses. But is this a
relaistic limitation? I thout you wanted to be able to edit the
configuration from various hosts. If this is only your setup, then an
sftp-based setup is probably more convinient.

Using a .htaccess file (or even better: an apache config snippet in
/etc/apache/conf.d )you can force authentication to get to this
directory. But then-again, you empower the apache user (www-data) to
configure and control asterisk, and thus if anybody manages to make your
web server execute an arbitrary script (e.g: can write to a .php file
under the wwwroot) they can make asterisk execute arbitrary code as
well.

The chmod command makes Asterisk's configuration world-writable. So
anybody with temporary write access to your system can again change
asterisk's configuration. This breaks a general sanity assumption that
only system users can write to the configuration. As a rule of thumb
such a chmod should generally be replaced by adding a certain user to a
certain group.

You also change the permissions to the asterisk reload script to 777.
Why does it need to be world-writable? This gives an attacker yet
another place to inject executable code.


In short: I still fail to see the atvantages of using phpconfig in your
settings.

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does the g.729 registration program work?

[Steven Critchfield]

On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote:
> There is an open source version of the license:
> 
> 
> You can view the licensing information at the following:
> 
> 
> more details can be found on http://www.voip-info.org

http://www.intel.com/software/products/ipp/speech_code_lin_eula.htm

And in section 3.E of the license above listed, you still must obtain
licenses from the patent holders. 

Also, arguably, section 8. removes the ability to use in most phone
systems. 

While it MAY be highly unlikely that you get prosecuted for using the
code on your home machine. But if you start to scale up to any
significant port usage, you may find yourself battling to not lose the
company on a simple licensable part.

Not to mention the fact that you will possibly get flamed for not having
gone the legal and community building route of buying from Digium.

> Steven Critchfield wrote:
> 
> >On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote:
> >  
> >
> >>I'm asking because I'm planning to install multiple machines from the
> >>same image and I need to know what file(s) I need to backup/restore to
> >>make sure I don't lose my licences in the process. The only options I
> >>can think of are:
> >>
> >>- There's a config file, though I've seen no mention of it
> >>- The actual binary shared library is modified
> >>- The system contacts Digium every time you start asterisk
> >>
> >>In the last case nothing is changed at all and I'm fine.
> >>
> >>
> >
> >It is based on a machine unique key created by querying your hardware.
> >You will not be able to share your licenses between machines. You will
> >need to buy licenses for each machine you deploy on.
> >  
> >
-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Fax on Asterisk

[Wiley Siler]

Perfect.  Thanks!  Found lots for incoming and that filled the gap for
out going.

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, February 25, 2005 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fax on Asterisk

On February 25, 2005 12:57 pm, Wiley Siler wrote:
> Is it possible to send an email to Asterisk and have it parse the 
> email or an attachment and send it out as fax?

Open up your web browser, go to www.google.com and enter "asterisk send
fax".  
That will get you info on how to send a .tiff file to a fax machine from
Asterisk.  The key you are likely missing is how to get email into a
pretty form.  look for apsfilter.  A little glue logic and it should be
pretty snappy for you.

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Fax on Asterisk

[Andrew Kohlsmith]

On February 25, 2005 12:57 pm, Wiley Siler wrote:
> Is it possible to send an email to Asterisk and have it parse the email
> or an attachment and send it out as fax?

Open up your web browser, go to www.google.com and enter "asterisk send fax".  
That will get you info on how to send a .tiff file to a fax machine from 
Asterisk.  The key you are likely missing is how to get email into a pretty 
form.  look for apsfilter.  A little glue logic and it should be pretty 
snappy for you.

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] "click to dial extension number" functionality ?

[marek cervenka]

By any web-user (ms explorer) to be able to call from a web-page to a 
certain number/extension connected to one specific asterisk.
maybe this php script help you (switch caller/called and modify Exten:)
--originate.php--



CALLER
CALLED




 $socket = fsockopen($astip,"5038", $errno, $errstr);
 fputs($socket, "Action: Login\r\n");
 fputs($socket, "UserName: $astmanager\r\n");
 fputs($socket, "Secret: $astpassword\r\n\r\n");
 fputs($socket, "Action: Originate\r\n");
 fputs($socket, "Channel: $tech/$caller\r\n");
 fputs($socket, "Context: $caller\r\n");
 fputs($socket, "Exten: $called\r\n");
 fputs($socket, "Priority: 1\r\n");
 fputs($socket, "Callerid: $caller\r\n\r\n");
 fputs($socket, "Action: Logoff\r\n\r\n");
 while (!feof($socket)) {
 $wrets .= fread($socket, 8192);
 }
 fclose($socket);
 echo "";
 echo <<";
}
?>

---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fax on Asterisk

[Wiley Siler]

Title: Fax on Asterisk






Is it possible to send an email to Asterisk and have it parse the email or an attachment and send it out as fax?


Thanks,

Wiley



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] VoIP/Asterisk presentation

[Ronald Hartmann]

Anyone able to get to these I am unable to get to them.


-Original Message-
From: Duane [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 25, 2005 10:14 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] VoIP/Asterisk presentation


On Sat, February 26, 2005 1:36, Ronald Hartmann said:
> Any chance you can share your presentation slides, or handouts etc.

Sure, but was only slides, no hand outs...

http://www.asterisk.net.au/voip%20in%203%20beers.pdf
http://www.asterisk.net.au/voip%20in%203%20beers.sxi
http://www.asterisk.net.au/voip%20in%203%20beers.ppt

Anyone is free to use the slides etc as long as both John Todd and I get
credit where credit is due etc...

-- 
Best regards,
 Duane

http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers

"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Directory config...

[Time Bandit]

> How do I config Asterisk so when the directory cmd is used, the name of
> the found entry comes from a pre-record gsm file instead of being spelled
> letter by letter?
If the user as recorded is name, this file will be used. When it's not
recorded, * will spell it.

Dial to your voicemail and navigate thru the menu to record your name.

hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does the g.729 registration program work?

[Mik Cheez]

There is an open source version of the license:

You can view the licensing information at the following:

more details can be found on http://www.voip-info.org
Steven Critchfield wrote:
On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote:
 

I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:
- There's a config file, though I've seen no mention of it
- The actual binary shared library is modified
- The system contacts Digium every time you start asterisk
In the last case nothing is changed at all and I'm fine.
   

It is based on a machine unique key created by querying your hardware.
You will not be able to share your licenses between machines. You will
need to buy licenses for each machine you deploy on.
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does the g.729 registration program work?

[Steven Critchfield]

On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote:
> I'm asking because I'm planning to install multiple machines from the
> same image and I need to know what file(s) I need to backup/restore to
> make sure I don't lose my licences in the process. The only options I
> can think of are:
> 
> - There's a config file, though I've seen no mention of it
> - The actual binary shared library is modified
> - The system contacts Digium every time you start asterisk
> 
> In the last case nothing is changed at all and I'm fine.

It is based on a machine unique key created by querying your hardware.
You will not be able to share your licenses between machines. You will
need to buy licenses for each machine you deploy on.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Speex transcoding for Cisco / Polycom

[I put the Who? in Mishehu]

Hi guys,
I have a weird problem, and I have encountered a few other people with 
the same issue.  The problem is this:

Whenever I make a call from my IAXy (g711ulaw) to my server, and then my 
server transcodes to speex and sends it to a remote asterisk server, 
audio is perfectly fine.  The same goes if I use Linphone with speex.  
However, whenever I use a Cisco 7960 SIP 6.2 or a Polycom IP 500 with 
SIP 1.3.1.0056, both using g711ulaw to my server, when it is transcoded 
to speex and a connection is made to the remote system, audio chops and 
I lose about 80% of all audio.  This has been tested on asterisk 1.0.3, 
1.0.4, 1.0.5, and CVS v1-0 from yesterday, all operate identically.  The 
problem works identically when I go to the remote and call to my server 
with it transcoding from the Polycoms to my server.

Hardware & software:
My server:  Dual Proc Xeon 2.8GHz/800Mhz FSB, 1 GB ECC Reg RAM, Intel 
SE7525GP2 motherboard, X101P (digium), 3ware RAID controller 9500.  
Slackware 10.0, speex 1.0.4, libogg-1.1-i486-1,  libvorbis-1.0.1-i486-1.

Remote server:  HP Proliant ML330 Xeon 3.06Ghz, Smartarray SCSI RAID, 
512MB ECC Reg RAM, TDM04b (digium).  Slackware 10.0, speex 1.0.4, 
libogg-1.1-i486-1,  libvorbis-1.0.1-i486-1.

Translation times are shown in asterisk on both machines as being in the 
vicinity of 28 to 45 ms from all other codecs to speex.

Ping times between machines:
24 packets transmitted, 24 received, 0% packet loss, time 23230ms
rtt min/avg/max/mdev = 15.770/53.820/164.989/43.473 ms
My codecs.conf:
[speex]
;0-10
quality => 3
;0-10
complexity => 4
; true / false
enhancement => true
; true / false
vad => false
; true / false
vbr => false
;0-10
abr_quality => 5
; true / false
abr => false
;0-10
vbr_quality => 5
; true / false
dtx => false
If anybody has any insight to this problem, it would be appreciated.  
(BTW, Speex 1.1.6 is just that, unstable...  it will even crash asterisk 
when you try to do "show translation recalc").

Thanks,
-Mishehu
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cascaded ringing

[Steven Critchfield]

On Fri, 2005-02-25 at 12:36 +, Julian J. M. wrote:
> You could add 
> exten => 1,2,Goto(context,2,2)
> 
> But I don't know what will happen when, after 5 secs, dial SIP/2 is
> executed again...

I think you are on the right track. If SIP considers ringing as busy
then you can cascade through your extensions doing the same goto if busy
logic.

call one comes in and is ringing/answered by sip/1
call two comes in and starts to ring Sip/1 and gets busy and rolls to
start ringing Sip/2 immediately. Sip/2 doesn't answer but when the wait
is over for Sip/2 normally, it sees Sip/2 is already "busy" and dials
Sip/3. You could then put voicemail at the end of Sip/3's timeout/busy
and you could catch the overflow.

Interesting way to handle it. I like it. I don't have a use for it, but
it is cool.


> On Fri, 25 Feb 2005 12:56:14 +0100, Elmar Haneke <[EMAIL PROTECTED]> wrote:
> > > exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL 
> > > PROTECTED]&Local/[EMAIL PROTECTED])
> > >
> > > [context]
> > > exten => 1,1,Dial(SIP/1)
> > > exten => 2,1,Wait(5)
> > > exten => 2,2,Dial(SIP/2)
> > > exten => 3,1,Wait(10)
> > > exten => 3,2,Dial(SIP/3)
> > >
> > > Basically, use the 'local' channel for your dial, then you can wait a
> > > bit before actually calling...
> > 
> > That's an good idea. How can I extend this to let SIP/2 ring
> > immediately if SIP/1 is busy?

-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

[Hecken, Guido]

You 're  right, there are some security issues using using sudoers
and system commands.
If the asterisk server is reachable from the outside over http or other
unsecure protocols, it would be really dangerous.
But in a trusty intranet-environment, where firewalls block every attempt
to access the asterisk server from the outside, this "solution" should be
save enough, even if nothing is really save enough ;-) .
If administrative access to any remote asterisk server is need, one could
use
vpn etc. to achieve secure connections.
Wouldn't this be sufficient?

>Consider using su-exec (and php in cgi) to run the configuration
>interface as the user asterisk or a special user.
Another idea:
Why not giving apache the right to execute only one shell command in
sudoers?
Something like 
apache CMD=(asterisk -r -x 'restart now')
could do the job.

Guido Hecken


> -Ursprüngliche Nachricht-
> Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
> Gesendet: Freitag, 25. Februar 2005 16:25
> An: asterisk-users@lists.digium.com
> Betreff: Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
> 
> Hi
> 
> On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote:
> > >Secondly, is the statement no.2 a line a need to change in a given
file?
> > You have to change/verify some settings in phpconfig_init.php .
> > Look for fakeuser=admin.
> > Set $reset_cmd = "./asterisk.reload";
> > Be shure, the script has write access in /etc/asterisk
> > Have something in your sudoers file (/etc/sudoers) like
> > apache ALL=(ALL)NOPASSWD: ALL
> 
> Why not simply run apache as root and be done with that?
> 
> Adding the following line to sudoers makes apache root-equivalent. Any
> attacher that is able to compromise apache gets your whole server.
> 
> > to allow apache execute system commands like asterisk -r -x 'restart
now'
> >
> > Another important file is the manager.conf in /etc/asterisk
> > [general]
> > enabled = yes
> > port = 5038
> > bindaddr = 0.0.0.0
> >
> > [admin]
> > secret = secret
> > permit = 192.168.0.0/255.255.255.0
> > read = system,call,log,verbose,command,agent,user
> > write = system,call,log,verbose,command,agent,user
> >
> > With these settings enabled, it should work.
> > Be aware, this is not a secure solution since allowing apache to execute
> > system-commands, and using the asterisk-web-dir (/var/www/html/asterisk)
> > without any further security actions like .htaccess file should only be
used
> > in trusted  environments like intranets.
> 
> Furthermore: anyone who can add arbitrary entries to your dialplan can
> use System to make apache run an arbitrary command. If you run asterisk
> as root (which you shouldn't) this gives the attacker a convinent root
> shell access. If not: it will only give the attacker the opportunity to
> run an arbitrary command as the asterisk user.
> 
> If you want to edit an arbiterary config file, use ssh. It is a
> well-tested, well understood and well-supported environment. Either edit
> directoly from the shell (you can't really bit vim ;-) ), or use an
> external X server and a more comfortable editor, or simply edit files
> via sftp.
> 
> > We can live with these restrictions. In the meanwhile we 're testing and
> > evaluating the complete asterisk configuration from within mysql.
> 
> Not much better, security-wise. I figure that the password to a mysql
> account with ability to write to the config (and specifically to the
> dialplan) will be availble in a certain location. So apache still has
> the ability to change the dialplan.
> 
> Consider using su-exec (and php in cgi) to run the configuration
> interface as the user asterisk or a special user.
> 
> --
> Tzafrir Cohen | New signature for new address and  |  VIM is
> http://tzafrir.org.il | new homepage   | a Mutt's
> [EMAIL PROTECTED] ||  best
> ICQ# 16849755 | Space reserved for other protocols | friend
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] "click to dial extension number" functionality ?

[adria vidal]

El 25/02/2005, a las 12:10, Terje Myhre escribió:
By any web-user (ms explorer) to be able to call from a web-page to a  
certain number/extension connected to one specific asterisk.

  
Almost as a web-based “auto-attendant” functionality. 
  
Hence:
1.   surf to the specific web-site
2.  enter the extension digits in a web-interface
3.  get connected – with in- and out-sound through the web-browser
  
Do anyone know what would be the simplest / best way to implement this  
functionality ?

We have developed something similar
http://www.asteriskspain.org/index.php? 
option=com_remository&Itemid=41&func=selectfolder&filecatid=1

you choose an extension put your phone number and asterisk make and  
bridge calls, it's easy to personalize if you want.

··
Adrià Vidal
xpreme.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Avaya Partner ACS3 and Asterisk

[jason]

Im sorry if this has been asked before as I couldnt seem to find it.

I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function and be able to dial a number like that.  I can get all
the asterisk stuff set up to do this, i just need to know if anyone knows
what setting to use in the partner system to make it bring up a dialtone
so i can dial.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] T.38 fax summary

[Steve Underwood]

Mark,
In the time it took to write all that you could probably have read up 
enough about T.38 to realise you were talking complete rubbish :-)

Regards,
Steve
Mark Eissler wrote:
On Feb 25, 2005, at 10:20 AM, Lee Howard wrote:
In a traditional analog fax you have modulated audio data, that is, 
the data stream is converted into an audio representation by the 
transmitter, and the receiver demodulates the audio stream to produce 
the data stream.  A lot of data gets packed into very small portions 
of audio, which is why fax over VoIP (T.38 is not VoIP, it is FoIP) 
is unreliable - any jitter will likely cause data loss.

There are no modulators in T.38.  So take the fax procedure, but 
instead remove the data modulation/demodulation part.  T.38 devices 
communicate raw data through the IP network, and the IP network is as 
good at communicating data as the PSTN is as good at communicating 
audio.  So if you could have a full T.38 delivery route from fax 
sender to fax receiver, the data never once gets converted into an 
audio signal - it doesn't need to be.

Sort of...but no. Fax requires a codec that supports the frequency 
spectrum of a POTS audio channel. Currently, that means that anything 
other than g.711 won't work since the other popular codecs achieve 
their efficiency by dumping frequencies humans can't hear (just like 
mp3). The problem isn't typically g.711 because that's the codec that 
is generally used by the digital telco world. A common problem when 
discussing g.711 often is packet size vs bandwidth limitations. T.38 
can alleviate this problem because it doesn't rely on a codec.

The bigger problem with faxing over VOIP is related to lost packets 
and timing issues (jitter). Lost packets are the death knell for fax 
because it isn't very tolerant of missing data. How do you complete an 
image with missing data??? AFAIK T.38 can't do anything to recover 
from packet loss...the fax machine needs to be tolerant of it. 
Ironically, ECM was introduced to recover from information loss when 
transmitting faxes over analog lines but ECM can actually cause 
problems when used with T.38. If you can turn ECM off that's the best 
thing to do when using T.38. Besides lost packets though if you have 
to consider packets arriving at weird timing intervals (jitter).

The fax machine needs to get its data in a steady stream. This is 
supposed to be a realtime transmission after all. While T.38 can 
absorb some of the problems triggered by latency and jitter, when the 
problem becomes too excessive it tanks just as quickly as faxing 
without T.38.

So with those barriers out of the way what is it that T.38 tries to 
accomplish? Instead of sending a fax over VOIP as a stream of sampled 
audio, the protocol intercepts the audio at the endpoints and 
packetizes it as blocks of data instead. The receiving gateway must 
know how to handle the data stream so it can convert the fax back into 
a T.30 fax data stream for POTS. During the session, progress is faked 
so that the two fax machines don't think the transmission has 
stopped...that's a crucial step because it takes time to convert and 
send/receive the fax reliably.

I think the best arsenal for faxing over VOIP today is to have a good 
broadband connection, g.711, and a fax machine where YOU can set the 
max transmission speed. Sadly, the last part seems to be missing quite 
often. I've noticed that HP actually mentions faxing over VOIP in the 
documentation for their 7410 all in one machine and, more importantly, 
they include support for changing transmission speeds. Way to go HP!

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Avaya Partner ACS3 and Asterisk

[jason]

Im sorry if this has been asked before as I couldnt seem to find it.

I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function and be able to dial a number like that.  I can get all
the asterisk stuff set up to do this, i just need to know if anyone knows
what setting to use in the partner system to make it bring up a dialtone
so i can dial.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] FW: Getting PHP Config to work?

[C. Tomlinson]

Hi,

Thanks for the batchfile type, it's a handy one.

I'm not editing over the internet, just local LAN for testing ATM. Protected
via firewall.

Would it not be fairly secure using a combination of the following:
.htaccess file
VPN?
https access?
Limit apache to only allow certain IP's?
And the public keys thing.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: 25 February 2005 15:49
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work?

On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote:
> Richard,
> 
> I have been using WinSCP to transfer files across easily without messing
> with FTP accounts. I had not found that feature, many thanks for pointing
it
> out :-D
> 
> I will definitely use this from now on until I find a better solution. Do
> you have an easy way to reload asterisk after changing the files? Have
putty
> open to do a reload? Or use the builtin terminal capabilities of WinSCP?

Basically you need to run one shell command. In linux I'd use:

  ssh [EMAIL PROTECTED] asterisk -rx reload

As this is a platform without native support of ssh, you can use the
command plink to get basically the same effect. Create a putty 
configuration called "rapidroot" to connect to [EMAIL PROTECTED] and use
something like

  plink rapidroot asterisk -rx reload

in a batch file. Or use [open]ssh from cygwin, if you're more comfortable
with
it.

You should use public-keys authentication to get better control .
Actually you can configure a certain public key so it will only allow
running one single command (asterisk -rx reload, in your case).

> 
> This is a great fix as my main machine is currently Windows. However I
would
> still like to get phpconfig working as it would be easier to use that
across
> the internet etc.

OVER THE INTERNET???

See my recent post on the previous thread about phpconfig. Allowing
phpconfig to do the same is quite insecure.

Also consider using mc from the shell.

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] FW: Getting PHP Config to work?

[C. Tomlinson]

Thanks for the chmod, it definitely needed that!
I didn't have to change the etc.sudoers file though. I'm running Debian, via
the great Xorcom rapid installation.

I didn't change the permit lines either as this is just  attesting box and
im not worried about security.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hecken, Guido
Sent: 25 February 2005 14:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work?

> ;
> [general]
> enabled = yes
> port = 5038
> bindaddr = 0.0.0.0
> 
> [admin]
> secret = secret
> ;deny=0.0.0.0/0.0.0.0
> ;permit=209.16.236.73/255.255.255.0

Do this in manger.conf, where xxx.xxx.xxx.0 represents your network:

[admin]
secret = secret
permit=xxx.xxx.xxx.0/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

To get the reload script running, you must allow apache (or the account
apache runs under) to execute scripts. Therefore use visudo (in Fedora) to
add the following line in /etc/sudoers
apache ALL=(ALL)NOPASSWD: ALL

To write the config files in /etc/asterisk with phpconfig.php you need to
give apache the rights to do so. A simple chmod -R a+w /etc/asterisk should
do the job.
I know, there are more secure methods to do this, but it works for us.

Guido Hecken

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort

[Alex G Robertson]

tim panton wrote:
[..]
Good luck with it.
I think I am lucky! ;-)
We resolved the problem changing the Mother board to one with an Intel chipset.
The first one had Via chipset.
1) At first, we changed the mother board by another with Intel chipset. This one
can set udma2 (or udma3) from BIOS. It is set up to udma2.
# ChannelBank 1 - T1
span=1,0,0,esf,b8zs
fxoks=1-24
# ChannelBank 2 (Empty) - T1
span=2,0,0,esf,b8zs
fxoks=25-48
# Empty - T1
span=3,0,0,esf,b8zs
fxoks=49-72
# Telco E1/PRI
span=4,0,0,ccs,hdb3
bchan=73-87
dchan=88
bchan=89-103
The messages "HDLC Bad FCS" and "HDLC Abort" stopped!
2) After that, we set up span 3 and 4 (3rd and 4th ports on the card)
as E1.
We are going to use 2 E1s/PRI (different Telcos) and 2 T1s (channelbank).
[...]
# Telco1
span=3,0,0,ccs,hdb3
bchan=...
dchan=...
bchan=...
# Telco2
span=4,0,0,ccs,hdb3
bchan=...
dchan=...
bchan=...
-> The jumpers were set to E1 and the channels were also correct.
-> After each modification it was necessary to reboot and take off the power
cable for modifications to take effect.
What timing/clock configuration do you recomend us? Zero for everyone?
 span=1,0,0,esf,b8zs # T1 ChannelBank1
 span=2,0,0,esf,b8zs # T1 ChannelBank2
 span=3,0,0,ccs,hdb3 # E1 Telco1
 span=4,0,0,ccs,hdb3 # E1 Telco2
By the way, could you explain in a better way what does these timing options (0,
1, 2) mean?

--
Alex G Robertson
NOC - Microlink
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] FW: Getting PHP Config to work?

[C. Tomlinson]

Thanks to all your help I have now got it working great.

I have written a quick howto which I plan to add to the wiki if people
approve?

Take a look at:
http://www.burntwires.com/asterisk/Install%20PHP%20Config.htm

(Please excuse the bloated html)

Please leave any feedback and then I will add to the wiki.

C

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson
Sent: 25 February 2005 13:32
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] FW: Getting PHP Config to work?

Hi,

I have been doing various testing with asterisk and its been going great.
However I am a bit feedup of using vi for editing configs, and would rather
do it from any machine on my LAN. I am running debian and * via xorcom rapid
on a test PC at the minute.

Hence phpconfig would be great, however I am having difficulty getting it to
work. I have searched the message boards and the wiki, and found nothing of
help for this problem :(

I have a full working apache/php setup (default install) and have added the
phpconfig files to the www dir, and they are accessible over the LAN. So far
so good.

I Can read the files fine.

However I cannot write any files, I get the error:

User: admindoes not have access to this feature.
Write failed! 

I tried messing with the CHMOD settings of the files but no joy.

My manager.conf looks like:

;
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[admin]
secret = secret
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.73/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

I can successfully telnet into the manager interface through shell on the
local machine and winxp machine on the LAN

I have moved asterisk.reload into /bin, and if I run it from the shell I get
a successful? Output:

pbx01:~# /bin/asterisk.reload
Asterisk Call Manager/1.0


Can anyone help? It is the same error the online example gives. Is it
something to do with specific admin rights in xorcom, or have I missed
something fundamentally wrong out? I have checked the php files and the
paths seem to be OK (default * installs)
I have a couple of ideas as to the problem:
-PHP needs something enabled e.g safemode?
-Xorcom has changed something phpconfig needs e.g * not running as root or
something?

Many Thanks,
C 




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] WebVMail Woirks but No Audio

[Richard J. Sears]

Hi Everyone - 

I have webvmail up and running, I can see the messages, forward them,
pretty much everything but listen to them.

Here is what I see in my logs:

192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET
/vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default
&password=12&msgid=&format=gsm&dontcasheme=4624.gsm HTTP/1.1"
200 9438 "-" "contype"


But the box at the bottom shows up as a broken link.

Any ideas...?


Thanks


**
Richard J. Sears
Vice President 
American Internet Services  

[EMAIL PROTECTED]
http://www.adnc.com

858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130


I fly because it releases my mind 
from the tyranny of petty things . . 


"Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching."

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] T.38 fax summary

[Mark Eissler]

On Feb 25, 2005, at 10:20 AM, Lee Howard wrote:
In a traditional analog fax you have modulated audio data, that is, 
the data stream is converted into an audio representation by the 
transmitter, and the receiver demodulates the audio stream to produce 
the data stream.  A lot of data gets packed into very small portions 
of audio, which is why fax over VoIP (T.38 is not VoIP, it is FoIP) is 
unreliable - any jitter will likely cause data loss.

There are no modulators in T.38.  So take the fax procedure, but 
instead remove the data modulation/demodulation part.  T.38 devices 
communicate raw data through the IP network, and the IP network is as 
good at communicating data as the PSTN is as good at communicating 
audio.  So if you could have a full T.38 delivery route from fax 
sender to fax receiver, the data never once gets converted into an 
audio signal - it doesn't need to be.

Sort of...but no. Fax requires a codec that supports the frequency 
spectrum of a POTS audio channel. Currently, that means that anything 
other than g.711 won't work since the other popular codecs achieve 
their efficiency by dumping frequencies humans can't hear (just like 
mp3). The problem isn't typically g.711 because that's the codec that 
is generally used by the digital telco world. A common problem when 
discussing g.711 often is packet size vs bandwidth limitations. T.38 
can alleviate this problem because it doesn't rely on a codec.

The bigger problem with faxing over VOIP is related to lost packets and 
timing issues (jitter). Lost packets are the death knell for fax 
because it isn't very tolerant of missing data. How do you complete an 
image with missing data??? AFAIK T.38 can't do anything to recover from 
packet loss...the fax machine needs to be tolerant of it. Ironically, 
ECM was introduced to recover from information loss when transmitting 
faxes over analog lines but ECM can actually cause problems when used 
with T.38. If you can turn ECM off that's the best thing to do when 
using T.38. Besides lost packets though if you have to consider packets 
arriving at weird timing intervals (jitter).

The fax machine needs to get its data in a steady stream. This is 
supposed to be a realtime transmission after all. While T.38 can absorb 
some of the problems triggered by latency and jitter, when the problem 
becomes too excessive it tanks just as quickly as faxing without T.38.

So with those barriers out of the way what is it that T.38 tries to 
accomplish? Instead of sending a fax over VOIP as a stream of sampled 
audio, the protocol intercepts the audio at the endpoints and 
packetizes it as blocks of data instead. The receiving gateway must 
know how to handle the data stream so it can convert the fax back into 
a T.30 fax data stream for POTS. During the session, progress is faked 
so that the two fax machines don't think the transmission has 
stopped...that's a crucial step because it takes time to convert and 
send/receive the fax reliably.

I think the best arsenal for faxing over VOIP today is to have a good 
broadband connection, g.711, and a fax machine where YOU can set the 
max transmission speed. Sadly, the last part seems to be missing quite 
often. I've noticed that HP actually mentions faxing over VOIP in the 
documentation for their 7410 all in one machine and, more importantly, 
they include support for changing transmission speeds. Way to go HP!

-mark
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Directory config...

[Francois Meehan]

Hi all,

How do I config Asterisk so when the directory cmd is used, the name of
the found entry comes from a pre-record gsm file instead of being spelled
letter by letter?

Regards,

Francois



Random Thought:
---
All of us failed to match our dreams of perfection. So I rate us on the basis 
of our splendid failure to do the impossible. - William Faulkner, 1897 - 1962
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] How does the g.729 registration program work?

[Martijn van Oosterhout]

I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:

- There's a config file, though I've seen no mention of it
- The actual binary shared library is modified
- The system contacts Digium every time you start asterisk

In the last case nothing is changed at all and I'm fine.

Thanks in advance,
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR writing incorrect data to pgsql tables

[James Bean]

> For MySQL and other glorified flat-file databases, you would 
> need to postprocess the data. You may feel more confident 
> skipping triggers and doing this anyway.
> 
> > So by that any calls that go out over the net using IAX to 
> the telco 
> > are considered digital and will report correctly?
> 
> Yes. You will probably be able to make the simple assumption 
> that if dstchannel ILIKE 'Zap/%' , you're going to have to 
> fudge it, otherwise it's correctly recorded.
> 

Thank you for your help sir it was very informative I am going to write
the trigger with my own rules for the database and see how I go :-)

James
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FW: Getting PHP Config to work?

[Tzafrir Cohen]

On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote:
> Richard,
> 
> I have been using WinSCP to transfer files across easily without messing
> with FTP accounts. I had not found that feature, many thanks for pointing it
> out :-D
> 
> I will definitely use this from now on until I find a better solution. Do
> you have an easy way to reload asterisk after changing the files? Have putty
> open to do a reload? Or use the builtin terminal capabilities of WinSCP?

Basically you need to run one shell command. In linux I'd use:

  ssh [EMAIL PROTECTED] asterisk -rx reload

As this is a platform without native support of ssh, you can use the
command plink to get basically the same effect. Create a putty 
configuration called "rapidroot" to connect to [EMAIL PROTECTED] and use
something like

  plink rapidroot asterisk -rx reload

in a batch file. Or use [open]ssh from cygwin, if you're more comfortable with
it.

You should use public-keys authentication to get better control .
Actually you can configure a certain public key so it will only allow
running one single command (asterisk -rx reload, in your case).

> 
> This is a great fix as my main machine is currently Windows. However I would
> still like to get phpconfig working as it would be easier to use that across
> the internet etc.

OVER THE INTERNET???

See my recent post on the previous thread about phpconfig. Allowing
phpconfig to do the same is quite insecure.

Also consider using mc from the shell.

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FW: Getting PHP Config to work?

[Tzafrir Cohen]

On Fri, Feb 25, 2005 at 03:15:34PM +0100, Michiel van Baak wrote:
> On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote:
> > Richard Folwell wrote:
> > >
> > >Look at WinSCP:
> > 
> > 
> > 
> > >It is (almost) worth installing Windows just to be able to use it. :-) 
> > >If anyone knows of anything similar that runs under Linux please 
> > >enlighten me!

* mc
* gnome's gnome-vfs
* kde's fish io-slave
* vim, as mentioned below.

And there bound to be others. You can also do this in the kernel level
using shfs.

> 
> scp
> This is installed together with the ssh binary.
> And if you are using vim/gvim you can do the following when
> in command mode
> :e proto://[EMAIL PROTECTED]//path/file
> see this vim tip:
> http://www.vim.org/tips/tip.php?tip_id=337

And did I mention that vim has syntax hilighting for asterisk extensions
file?

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Fedora Core 3?

[Race Vanderdecken]

I am developing voicemail and SIP and RAIDUS code for Asterisk Code on
the Fedora Core 3 and having no problems.

I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6 I could understand.)

It has been running for 2 weeks. It compiles fast and easy and no
complaints from asterisk CVS from 2 weeks ago.

Race "The Tyrant" Vanderdecken



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of j
Sent: Friday, February 25, 2005 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fedora Core 3?

I use FC3 on all our servers including 3 * servers.

  I have absolutely no issues what so ever.
  You do NOT need the kernel source RPM (which I don't even think exists
anymore) as they've changed how they set up the kernel RPMs somewhere
after FC1.
 
  The source rpm from FC1 (which is a bit old 2.6.5 or something) is if
you actually want to compile your own kernel. 
  The regular kernel rpms now come with all the headers and development
stuff included.

  You should be able to install the kernel rpm and compile zaptel right
away.

  do an rpm -ql kernel | less to check out the contents. They have
header files all over the place ;)

  Cheers.

j

On Fri, 2005-02-25 at 08:15 -0500, Darren Ellis wrote:
> Rich Adamson wrote:
> 
> >Is there any reason to avoid * on Fedora Core 3 at this time? 
> >Have most/all of the issues been resolved now?
> >
> >  
> >
> Rich,
> 
> Both my Asterisk servers run FC3.  The only issue I ran into was the 
> change in RPMs for the source.  FC doesn't distribute the 
> "kernel-source" RPM any more.  You need to get the SRPM.  No big deal,

> and it's documented on the Fedora Core website.
> 
> My servers are not in production, however.  I'm still working out 
> configuration issues.  Feel free to contact me off-list if I can be of

> further assistance.
> 
> Darren
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
j <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk and 723,729

[Race Vanderdecken]

The Cheapest way is to purchase 2
licenses, or in multiples of 2 If you need more, from Digium.

 

You will be beating a dead horse and a
dead carriage and a dead driver if you try to get around G729 licensing. You only
need a license for each answer and originate session that uses g.729 when
talking with asterisk itself, not the pass through conversations. 

 

Use the Erlang calculator, http://www.erlang.com/calculator/lipb/,
to determine the number of licenses you need. 

 

You DON’T need a license for every subscriber/users,
just for the number of users that will be talking with Asterisk via voicemail
and prompts.

 

G.729 from phone to phone passes directly
through asterisk and does not require a license.

 

Race “The Tyrant” Vanderdecken

 

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne
Sent: Friday, February 25, 2005
5:43 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk
and 723,729

 



has any one implemented asterisk
with 723 and 729 codecs, what is the cheapest way.





is there a limitation in the open
723 implementation ??





 








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] FW: Getting PHP Config to work?

[C. Tomlinson]

Hi,

I'm not sure the way to change it, but when I d/l it from 

http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig/cls_phpconfig.ph
p?login=2&asterisksess=5c8e63576772790cfc2e1dbce354e04d

I had read about the problem with fget's, but presumed this change was the
correct one. However it looks like my skim reading got the better of me!

I am writing up an installation guide now.

C

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: 25 February 2005 15:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work?

> Some time ago, I had the same probs with phpconfig and had to search and
> google quite a long time to get it running. Since our systems are now
> running fine with phpconfig, I simply forgot the above fgetc/fgets issue.
> Therefore...
> A wonderful place for all this would be the wiki ;-)
Better yet, update the CVS with the correction.

How would I go about that ?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: FRS and GMRS via *

[Michael B. Murdock]

There are GMRS radios that support frequency splits... I dont think FRS
does.

-- Mike

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

Sent: Friday, February 25, 2005 9:10 AM
Subject: Re: [Asterisk-Users] Re: FRS and GMRS via *


> > Would plugging into the headphone jack with a phone-patch-type device
> > be considered a modification for radios with vox capability?
> ah ah so do ' phone-patch-type device'  interface
> via the to frs/gmrs 2 way radios via the mic jack ?
> can someone that know this stuff point out a few urls of the phone patches
?
>
> is there such thing as frs/gmrs repeater that can send/receive on
different
> frequencies at the same time to acheive a duplex conversation ?
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CDR writing incorrect data to pgsql tables

[Peter Corlett]

James Bean <[EMAIL PROTECTED]> wrote:
[...]
> Is the kludge done at the software side when the data is pulled out
> for accounting and being under say 45 seconds is a no answer or
> busy? Or is there a tweak that can be done at the database itself?

Since you're using PostgreSQL, you can use a trigger to mangle the
data before it hits the database. In fact, there's no reason why you
couldn't log to a view rather than a table (but again, you will need a
trigger for the actual INSERT.)

For MySQL and other glorified flat-file databases, you would need to
postprocess the data. You may feel more confident skipping triggers
and doing this anyway.

> So by that any calls that go out over the net using IAX to the telco
> are considered digital and will report correctly?

Yes. You will probably be able to make the simple assumption that if
dstchannel ILIKE 'Zap/%' , you're going to have to fudge it, otherwise
it's correctly recorded.

-- 
The intuitive mind is a sacred gift and the rational mind is a faithful
servant. We have created a society that honors the servant and has forgotten
the gift.
- Albert Einstein
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Vonage <---> Asterisk Complete Config

[Jay Milk]

Must have missed a few messages :)  Vonage always allowed this on
"softphone" lines.  Those are $10/month with metered usage (100 min
included).  They also require a "hardline" (ATA) as the primary line on
the account.  It's a working crutch for those folks who need a DID in a
rate-center only vonage offers -- but that number, thankfully, is
decreasing.

> -Original Message-
> From: Randy Johnson [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 25, 2005 6:27 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Vonage <---> Asterisk Complete Config
> 
> 
> I thought Vonage did not allow this?
> 
> 
> -Randy
> 
> 
> Nitesh Divecha wrote:
> 
> >Hello Asterisk Users,
> >
> >After Brain storming for couple of hours, days, and weeks, 
> finally got 
> >Asterisk to work with Vonage for Inbound and Outbound calls.
> >
> >Requirement: -
> >1) Vonage Softphone account
> >2) Asterisk
> >3) Couple of SIP Phones
> >
> >Here is my sip.conf
> >
> >[general]
> >port = 5060   ; Port to bind to (SIP is 5060)
> >bindaddr = <>; Address to bind to (all 
> addresses on machine)
> >context=incoming
> >disallow=all
> >allow=ulaw
> >allow=alaw
> >allow=g729
> >allow=g723
> >externip=<>
> >localnet=<>
> >localmask=<>
> >nat=yes
> >
> >register=<>:[EMAIL PROTECTED]:5061/202
> >
> >[vonage-out]
> >username=<>
> >type=friend
> >secret=<>
> >port=5061
> >nat=yes
> >host=sphone.vopr.vonage.net
> >fromuser=<>
> >fromdomain=sphone.vopr.vonage.net
> >dtmfmode=rfc2833
> >auth=md5
> >
> >[vonage202]
> >username=<>
> >type=friend
> >secret=<>
> >port=5061
> >nat=yes
> >insecure=very
> >host=sphone.vopr.vonage.net
> >fromuser=<>
> >fromdomain=sphone.vopr.vonage.net
> >dtmfmode=inband
> >context=from-pstn
> >canreinvite=no
> >auth=md5
> >
> >Here is my extension.conf
> >
> >[ext-did]
> >exten => <,1,Goto(ext-local,202,1)
> >or 
> >exten => <>,1,Goto(aa_1,s,1) If you are sending 
> the call to IVR.
> >
> >For some this configuration might vary as my Asterisk is behind NAT.
> >
> >Asterisk Rocks!!! Enjoy
> >
> >Many thanks to Jay & Dean
> >
> >Neel
> >
> >
> >
> >___
> >Asterisk-Users mailing list
> >Asterisk-Users@lists.digium.com 
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >  
> >
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/aster> isk-users
> To 
> UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ATA that actually work with T.38

[Mark Eissler]

On Feb 25, 2005, at 7:55 AM, Steve Underwood wrote:
If you understand what T.38 is you will understand which problems it 
addresses (summary: it is important for solving some problems, but 
nothing solves them all). Most people who post about T.38 don't 
actually have much of a clue about it.
I think the biggest hurdle still for T.38 is lost packets and timing 
issues. In other words, the realtime-ness (?) of it is a huge problem. 
IMHO the whole thing's a bust until we all get QoS across the public 
network. And let's face it, if you have a private IP network with QoS 
you really don't need T.38. So I'm a bit lost as to how T.38 is really 
a solution to much of anything at this point yet the hype would have 
one conclude otherwise.

As for Asterisk not having to know much about T.38...well, that's only 
true if the only support that will be available (on the Asterisk end) 
is via an analog adapter that supports T.38. If you want to hookup a 
fax machine to a port on a channel bank or a zap card then you're going 
to be out of luck unless the zaptel driver supports T.38.

-mark
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] ATA that actually work with T.38

[Andreas Sikkema]

[EMAIL PROTECTED] wrote:

> For T.38 passthrough between RTP channels it doesn't need to know a
> great deal. There are some pitfalls, though, due to dumbness
> in the T.38
> spec.
> 
> Are you actually working on this?

Yes, well, with a lot of other things, so progress is erratic. I've 
got to solve some other problems first, but Asterisk T.38 pass 
through is the next major issue.

-- 
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Working SIP phone for linux and windows

[Time Bandit]

> I have yet to discover a software package that would both register and
> have ulaw codec. The SIP communicator (Java) came closest to usable,
> but didn't have the ulaw codec working.  What do you use for
> communications?
for SIP you can use X-Lite :
http://www.xten.com/index.php?menu=products&smenu=download

I think there's also a Linux version in beta, but I don't have the link near me.

If you want an IAX softphone with ulaw, I've done one :
http://www.marccharbonneau.com/asterisk/mediaxphone.php

hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP Errors

[Race Vanderdecken]

Hmmm,

Looking directly at the .../channels/chan_sip.c code does not get any
clues.


Switch( resp )
...
...

   case 480: /* Temporarily Unavailable */
   case 404: /* Not Found */
   case 410: /* Gone */
   case 400: /* Bad Request */
   case 500: /* Server error */
   case 503: /* Service Unavailable */
  if (owner)
  ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
  break;

Basically the code says that "something happened that we have not
written code to deal with the problem so lump it in with other things we
don't handle and tell the SIP device on the other then, "Doh!"

I found this reference from the "Gods and Generals" at Cisco:
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/relnotes/
solrelnt.htm
+++
"Problem: Server Internal Error might be returned in response to a
REGISTER request (CSCds02480)

Problem Description: Occasionally, the Cisco SIP Proxy Server returns a
"500 Server Internal Error" response to a REGISTER request. This problem
occurs primarily during periods of heavy CPU loads and receiving
REGISTER requests at a rate equal to or greater than 10 per second.
Also, this problem is more likely to occur when running a server farm
because the registration information is being updated on multiple
machines. This condition is temporary.

Recommended Action: Reissue the SIP REGISTER request."


Like I said, the SIP server is responding with "Doh!" Any more clues as
to when this happens?

Race "The Tyrant" Vanderdecken


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C.
Fertig
Sent: Friday, February 25, 2005 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Errors
Importance: High

Can someone explain what this error is? 

-- Got SIP response 500 "Server Internal Error - Invalid CSEQ number"
back from 209.xxx.xxx.xxx

How do I fix this?

 
 
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

[Tzafrir Cohen]

Hi

On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote:
> >Secondly, is the statement no.2 a line a need to change in a given file?
> You have to change/verify some settings in phpconfig_init.php . 
> Look for fakeuser=admin.
> Set $reset_cmd = "./asterisk.reload";
> Be shure, the script has write access in /etc/asterisk
> Have something in your sudoers file (/etc/sudoers) like
> apache ALL=(ALL)NOPASSWD: ALL

Why not simply run apache as root and be done with that?

Adding the following line to sudoers makes apache root-equivalent. Any
attacher that is able to compromise apache gets your whole server.

> to allow apache execute system commands like asterisk -r -x 'restart now'
> 
> Another important file is the manager.conf in /etc/asterisk
> [general]
> enabled = yes
> port = 5038
> bindaddr = 0.0.0.0
> 
> [admin]
> secret = secret
> permit = 192.168.0.0/255.255.255.0
> read = system,call,log,verbose,command,agent,user
> write = system,call,log,verbose,command,agent,user
> 
> With these settings enabled, it should work.
> Be aware, this is not a secure solution since allowing apache to execute
> system-commands, and using the asterisk-web-dir (/var/www/html/asterisk)
> without any further security actions like .htaccess file should only be used
> in trusted  environments like intranets.

Furthermore: anyone who can add arbitrary entries to your dialplan can
use System to make apache run an arbitrary command. If you run asterisk
as root (which you shouldn't) this gives the attacker a convinent root
shell access. If not: it will only give the attacker the opportunity to
run an arbitrary command as the asterisk user.

If you want to edit an arbiterary config file, use ssh. It is a
well-tested, well understood and well-supported environment. Either edit
directoly from the shell (you can't really bit vim ;-) ), or use an
external X server and a more comfortable editor, or simply edit files
via sftp.

> We can live with these restrictions. In the meanwhile we 're testing and
> evaluating the complete asterisk configuration from within mysql.

Not much better, security-wise. I figure that the password to a mysql
account with ability to write to the config (and specifically to the
dialplan) will be availble in a certain location. So apache still has
the ability to change the dialplan.

Consider using su-exec (and php in cgi) to run the configuration
interface as the user asterisk or a special user.

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users