Re: [asterisk-users] Mp3 for IVR prompts

2009-10-11 Thread Tilghman Lesher
On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote:
> Can I convert my .WAV IVR greetings, MOH and other recordings into G729
> format to prevent transcoding and hence CPU usage ?

You can do it with a one-time conversion.  This will require a single g729
license:

*CLI> file convert foo.wav foo.g729

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image

2009-10-11 Thread James Stocks

On 3 Oct 2009, at 20:38, James Stocks wrote:

> On 3 Oct 2009, at 16:37, Jonathan Thurman wrote:
>
>> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks 
>> wrote:
>>> Hi everyone,
>>>
>>> I hope someone can help me with a problem I'm having with Cisco 7940
>>> phones on the SIP 8.12 image.  When I place a call from one of the
>>> handsets, the call proceeds as normal for 20 seconds and is then
>>> terminated by Asterisk (1.4.26.2):
>>>
>>
>> We are runing 08-12-00 on 7940/60s just fine (Asterisk 1.6.1.1), and
>> have been for a while.
>>
>>>
>>> As far as I can tell, the 'a=silenceSupp:off - - - -' header is not
>>> accepted by the 7940, which seems like a bug in the SIP image to me.
>>> However, I can't find a way to report this problem to Cisco  
>>> without a
>>> support contract (which I do not have).  Reverting to version 7.5
>>> fixes the problem, but it is still present in 8.11.  The problem is
>>> not present if the PSTN initiates the call, nor is it present if I
>>> allow the handsets to reinvite each other.  Here's the sip.conf
>>> snippet if it helps:
>>>
>>
>> That all looks fine to me.  What do your SIPDefault.cnf and
>> SIP.cnf files look like?
>>
>> -Jonathan
>
> Hi Jonathan,
>
> Thanks for your reply.  Here's the two files, SIPDefault.cnf:
>
>
> # Image Version
> image_version: "P0S3-8-12-00"
>
> # Proxy Server
> proxy1_address: "pabx.spruce" # IP address here alternatively
>
> # Proxy Registration (0-disable (default), 1-enable)
> proxy_register: "1"
>
> # Setting for Message
> messages_uri: "222"
>
> # Time Server
> sntp_mode: "unicast"
> sntp_server: "snakebite.spruce" # IP address here alternatively
> time_zone: "GMT"
> dst_offset: "1"
> dst_start_month: "March"
> dst_start_day: ""
> dst_start_day_of_week: "Sun"
> dst_start_week_of_month: "4"
> dst_start_time: "02"
> dst_stop_month: "Oct"
> dst_stop_day: ""
> dst_stop_day_of_week: "Sunday"
> dst_stop_week_of_month: "4"
> dst_stop_time: "2"
> dst_auto_adjust: "1"
> date_format: "D/M/Y"
>
> # XML file that specifies the dialplan desired
> dial_template: "dialplan"
>
> #Time Format (0-12hr, 1-24hr [default])
> time_format_24hr: "1"
>
> # URL for external Phone Services
> services_url: "http://pabx.spruce/openxmldir/PhoneUI/index.php"; # IP
> address here alternatively
>
> # URL for external Directory location
> directory_url: "http://pabx.spruce/openxmldir/PhoneUI/index.php"; # IP
> address here alternatively
>
> # URL for branding logo
> logo_url: "http://pabx.spruce/cisco/asterisk.bmp"; # IP address here
> alternatively
>
>
> and SIP.cnf:
>
>
> # Image Version
> image_version: "P0S3-8-12-00"
> phone_label: " "
>
> # Line 1 appearance
> line1_displayname: "James"
> line1_shortname:"200 James"
> line1_name: 200
> line1_authname: "200"
> line1_password: "*removed*"
>
> # Line 2 appearance
> line2_displayname: "Work"
> line2_shortname: "206 Work"
> line2_name: 206
> line2_authname: "206"
> line2_password: "*removed*"
>
> # Line 3 appearance
> line3_displayname: ""
> line3_shortname: ""
> line3_name: UNPROVISIONED
> line3_authname: "UNPROVISIONED"
> line3_password: "UNPROVISIONED"
>
> # Line 4 appearance
> line4_displayname: ""
> line4_shortname: ""
> line4_name: UNPROVISIONED
> line4_authname: "UNPROVISIONED"
> line4_password: "UNPROVISIONED"
>
> # Line 5 appearance
> line5_displayname: ""
> line5_shortname: ""
> line5_name: UNPROVISIONED
> line5_authname: "UNPROVISIONED"
> line5_password: "UNPROVISIONED"
>
> # Line 6 appearance
> line6_displayname: ""
> line6_shortname: ""
> line6_name: UNPROVISIONED
> line6_authname: "UNPROVISIONED"
> line6_password: "UNPROVISIONED"
>
> # Phone Prompt (The prompt that will be displayed on console and  
> telnet)
> phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP
> Phone)
>
> # Phone Password (Password to be used for console or telnet login)
> phone_password: "*removed*" ; Limited to 31 characters (Default -  
> cisco)
>
> # User classifcation used when Registering [ none(default), phone,  
> ip ]
> user_info: none

OK.  For anyone finding this thread, the problem exists in Asterisk  
1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem.

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Re: [asterisk-users] Mp3 for IVR prompts

2009-10-11 Thread Michael Graves
On Sun, 11 Oct 2009 09:10:27 -0500, Tilghman Lesher wrote:

>On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote:
>> Can I convert my .WAV IVR greetings, MOH and other recordings into G729
>> format to prevent transcoding and hence CPU usage ?
>
>You can do it with a one-time conversion.  This will require a single g729
>license:
>
>*CLI> file convert foo.wav foo.g729
>

There are embedded Astrisk distros like Astlinux that target small form
factor hardware. They have distributed the standard prompts in various
encodings for quite some time. I think that they may have been amonst
the first to do so. Platforms like the Soekris Net4801 could only
sustain a couple of G.729 transcodes, making native prompts a practical
necessity.

Michael
--
Michael Graves
mgravesmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves




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Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image

2009-10-11 Thread Jonathan Thurman
On Sun, Oct 11, 2009 at 8:03 AM, James Stocks  wrote:
> OK.  For anyone finding this thread, the problem exists in Asterisk
> 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem.

Sorry, I lost your last response in my inbox...  Your phone configs
look fine.  The only thing that we do differently is disable VAD on
the phones.

Never used 1.4, only the 1.6 branch.  Glad to see that you got it working.

-Jonathan

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Re: [asterisk-users] Zaptel problems on SuSE 9.3

2009-10-11 Thread Angus Asterisk

- Original Message - 
From: "Philipp Kempgen" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, October 05, 2009 10:39 PM
Subject: Re: [asterisk-users] Zaptel problems on SuSE 9.3


Angus Asterisk schrieb:

> It seems that the zaptel startup script does not work.  I noticed at 
> startup
> the line:
> /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or
> directory
>

Just some feedback which might be helpful.

The VIA box I am running on has an internal modem and I think that might 
have had a resource clash with the Digium board.  So I disabled that plus 
other hardware devices I din't need in the bios.

The zaptel startup script doesn't seem to work on suse so I added:
modeprobe wctdm
ztcfg -vvv
asterisk

in /etc/init.d/boot.local

I think Suse is probably a bit over the top for asterisk.  Too much stuff 
running.  I might look at some really cut down linux distros.

What do other people use?


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[asterisk-users] Call Recording and Posting

2009-10-11 Thread Dan Journo
Hello,

 

I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.

 

Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp directory for new recordings?

 

I ask because I don't have any experience in Linux programming, so I
won't be able to create a monitoring program on my own.

 

Many thanks

Dan Journo

 

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Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-11 Thread Joseph
I just double checked the setting of the remote asterisk and it has the same 
setting as mine.
Sip.conf has in Global:
dtmfmode = rfc2833
individual extension has no dtmf setting at all, so global setting take 
precedence.

All units Linksys, Sipura have 
DTMF Tx Method: Auto

Linksys has an additional setting:
DTMF Tx Mode: Strict

My asterisk is using old Sipura units and dtmf tones to access voicemail are 
recognized.
The remote asterisk is using newer Linksys units and dtmf to voicemail does not 
work, the phone hangs up.  

The strange part is:
PSTN --> Asterisk (voicemail access) works OK on both sytemes.
Asterisk (w/Linksys) --> Asterisk (w/Sipura) to Voicemail works OK
Asterisk (w/Linksys) --> Asterisk (w/Linksys) to Voicemail DOES NOT work
Asterisk (w/Sipura) -->  Asterisk (w/Linksys) to Voicemail DOES NOT work

So it seems to me the Linksys units don't work as they suppose to.

--
Joseph

On 10/11/09 01:27, Ivan Stepaniuk wrote:
>Joseph wrote:
>> I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
>> The other Asterisk Linksys is set dtmf = auto
>If understand correctly, you have two asterisk servers and when you dial
>from one the other, DTMF is not recognized. I also asume you are using
>SIP to connect them as you mentioned dtmfmode. In any case, this should
>be set to the same value on both sides, both rfc2833, or both info. You
>don't wand inband and auto is just rfc2833 with automatic inband fallback.
>
>--
>Iv?n Stepaniuk
>Alba Fot?nica S. L.
>www.albafotonica.com

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[asterisk-users] Grandstream 2010

2009-10-11 Thread Cary Fitch
The Grandstream 286s automatically re register when a connection is
restored.

Our Grandstream 2010s don't.  Does anyone know of a setting that makes them
reregister?  I has tweaked "Watchdog timer" and anything that looked
promising.

Cary Fitch
Affordable Telecom


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Re: [asterisk-users] Call Recording and Posting

2009-10-11 Thread Elliot Otchet
Dan,

You can do this directly in the dialplan.  See the System command.  It allows 
you to call any program on the system (ftp, scp, mv, etc).  Keep in mind that 
depending on the volume of calls you're handling, you might run into I/O issues 
on the disk side.  If you're talking about a machine under enough load, you 
might need another alternative.

-Elliot

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, October 11, 2009 7:15 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Recording and Posting

Hello,

I'm working on a call recording solution. I would like recordings to either be 
automatically uploaded via FTP, or posted to a URL for processing by our main 
server.

Is Asterisk capable of doing this or will I have to create a separate 
application that monitors a temp directory for new recordings?

I ask because I don't have any experience in Linux programming, so I won't be 
able to create a monitoring program on my own.

Many thanks
Dan Journo



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Re: [asterisk-users] Call Recording and Posting

2009-10-11 Thread Steve Edwards
On Mon, 12 Oct 2009, Dan Journo wrote:

> I'm working on a call recording solution. I would like recordings to
> either be automatically uploaded via FTP, or posted to a URL for
> processing by our main server.
>
> Is Asterisk capable of doing this or will I have to create a separate
> application that monitors a temp directory for new recordings?
>
> I ask because I don't have any experience in Linux programming, so I
> won't be able to create a monitoring program on my own.

There is no built in facility -- but there are all the "parts."

There is the curl() application, but I don't know if it exposes enough 
"curl" to upload files.

There is the system() application which will let you execute any command 
line you can construct.

There is the agi() application which lets an external program interact 
with the dialplan.

Monitoring a temp directory with an external program would be the "worst" 
way.

Personally, I would wrap up the entire "call recording solution" in an AGI 
so you have a full featured language (my preference is C) and can hide all 
the ugly details and keep your dialplan simple and maintainable.

I've done these kind of applications where either a "control file" needed 
to be written and uploaded with the recording or a database needed 
updating. Both of these can get ugly in a dialplan.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4

2009-10-11 Thread Jeff Johnson

We are running Asterisk 1.4 and need some help to determine how (if)  *
supports 3 party warm transfers.  I've searched quite a bit  and all I
can find is information on "attended transfers".  What we are looking
for is: (1) external inbound call A comes to * extension B, caller A is
placed on hold and extension B calls external third party C.  After
explaining caller A issue to Party C, Ext B brings Caller A onto the
call and introduces A to C.  After the into, ext B then drops off the
call while A & C continue the call.  Any help would be appreciated.

Thanks Much, 

Jeff Johnson




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Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4

2009-10-11 Thread Lee, John (Sydney)
I don't think this can be done.
In your scenario, B is effectively the host and if B drops the line, both A and 
C will be dropped off as well.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Johnson
Sent: Monday, 12 October 2009 2:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4


We are running Asterisk 1.4 and need some help to determine how (if)  * 
supports 3 party warm transfers.  I've searched quite a bit  and all I can find 
is information on "attended transfers".  What we are looking for is: (1) 
external inbound call A comes to * extension B, caller A is placed on hold and 
extension B calls external third party C.  After explaining caller A issue to 
Party C, Ext B brings Caller A onto the call and introduces A to C.  After the 
into, ext B then drops off the call while A & C continue the call.  Any help 
would be appreciated.
Thanks Much, 
Jeff Johnson 
This email and any attached files are confidential and intended solely for the 
intended recipient(s). If you are not the named recipient you should not read, 
distribute, copy or alter this email. Any views or opinions expressed in this 
email are those of the author and do not represent those of NeturallySpeaking, 
LLC. 

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