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sandeep.s at briotelec...
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 Posted: Thu Jan 17, 2008 12:33 am Post subject: [asterisk-users] asterisk-users Digest, Vol 42, Issue 51 |
 |
|
hi all,
how to set the caller id facility for
the TDM400p card.
Please help me
thanks,
sandeep.s
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 15, 2008 3:09 PM
Subject: asterisk-users Digest, Vol 42, Issue 51
| Quote: | Send asterisk-users mailing list submissions to
asterisk-users at lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request at lists.digium.com
You can reach the person managing the list at
asterisk-users-owner at lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."
Today's Topics:
1. Re: app_voicemail for spanish (Andrew Joakimsen)
2. SVN servers down for maintenance (Russell Bryant)
3. Re: Asterisk 1.4.17 crashing more (Steve Totaro)
4. Zaptel 1.2.23 and 1.4.8 released (The Asterisk Development Team)
5. Re: AGISTATUS is SUCCESS even though my PHP script returned
-1 (Matt Riddell)
6. Re: Video Call and Asterisk (Matt Riddell)
7. Re: app_voicemail for spanish (Anton Krall)
8. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Steve Totaro)
9. Re: Asterisk RFC2833 to SIP INFO DTMF conversion erros. (Mayur)
10. Re: AGISTATUS is SUCCESS even though my PHP script returned
-1 (Steve Edwards)
11. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Tzafrir Cohen)
12. Park() help, extension not heard (Rob)
13. Re: AGISTATUS is SUCCESS even though my PHP script returned
-1 (Brian Hutchinson)
14. Re: Asterisk 1.4.17 crashing more (Brian Hutchinson)
15. Re: app_voicemail for spanish (Andrew Joakimsen)
16. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Andrew Joakimsen)
17. Re: Park() help, extension not heard (Rob)
18. pickupchan without bristuffed version? (Stefan Guenther)
19. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Bruce McAlister)
20. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Thomas Kenyon)
21. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Andrew Joakimsen)
22. Re: G.729 pre-compiled binaries and Asterisk 1.2.x.
(Thomas Kenyon)
----------------------------------------------------------------------
Message: 1
Date: Mon, 14 Jan 2008 18:57:34 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] app_voicemail for spanish
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801141557o7c84fa5ah3545781a978e230e at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
The language support is supposed to be there I know I've played with
it and there are at least SOME grammatical changes (don't recall which
right now)
But if further language support is needed you should file a bugreport.
On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote:
| Quote: | Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying "trabajo mensjes"
would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can
handle singular and plural (mensaje vs. mensajes)?
Anton
_______________________________________________
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|
------------------------------
Message: 2
Date: Mon, 14 Jan 2008 17:59:51 -0600
From: Russell Bryant <russell at digium.com>
Subject: [asterisk-users] SVN servers down for maintenance
To: undisclosed-recipients:;
Message-ID: <478BF777.5030903 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
The Digium svn servers are down, and will likely be down for the rest of
the
evening, as I perform some system maintenance. I apologize for any
inconvenience that this may cause.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
------------------------------
Message: 3
Date: Mon, 14 Jan 2008 19:03:21 -0500
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Subject: Re: [asterisk-users] Asterisk 1.4.17 crashing more
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<ea18e54a0801141603i2569a9d2i58011e5000fcfec at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Jan 14, 2008 6:23 PM, Abdul <abdul_zu at yahoo.com> wrote:
| Quote: | Hi All,
We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one
day it stop to response to the SIP Clinets so they cannot make call or
register. But safe_asterisk not restarting it back because asterisk
running
without any response to the sip clients.
When we try to do 'core show channels' using Manager it returns only
Action: Command
Command: show channels
That time asterisk not responding anything for clients for registration
either for invitation.
Please advice us how we can fix this issue.
|
Upgrade to Asterisk 1.2.X unless you need the features in 1.4.
Thanks,
Steve Totaro
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Message: 4
Date: Mon, 14 Jan 2008 18:03:28 -0600
From: The Asterisk Development Team <asteriskteam at digium.com>
Subject: [asterisk-users] Zaptel 1.2.23 and 1.4.8 released
To: undisclosed-recipients:;
Message-ID: <478BF850.7020702 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
The Asterisk.org development team has released Zaptel versions 1.2.23 and
1.4.8.
These releases contain a number of bug fixes as well as new features,
including:
* New and greatly improved fxotune utility
-
http://lists.digium.com/pipermail/asterisk-users/2008-January/203778.html
* Full support for new Digium cards, TE120P, TE121P, TE122P
* DTMF generator updates allow tones to be generated at runtime, as well
as support for a DTMF "twist", on a per-zone basis. The tones for
Brazil
have been updated to include a 2 dB DTMF twist.
These releases are available for immediate download from
http://downloads.digium.com/.
Thank you for your support!
------------------------------
Message: 5
Date: Tue, 15 Jan 2008 13:21:56 +1300
From: Matt Riddell <matt at venturevoip.com>
Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP
script returned -1
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478BFCA4.4010606 at venturevoip.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Brian Hutchinson wrote:
| Hi,
|
| Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter
| what my script returns (0 or -1), AGISTATUS always appears to be 0 =
| SUCCESS.
|
| I was wanting my script to be able to return a value to the dialplan and
| then test AGISTATUS but it looks like I'm going down the wrong path.
|
| Any suggestions?
Why don't you just set a variable from the AGI and then test for it in
the dialplan
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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------------------------------
Message: 6
Date: Tue, 15 Jan 2008 13:26:37 +1300
From: Matt Riddell <matt at venturevoip.com>
Subject: Re: [asterisk-users] Video Call and Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478BFDBD.4080701 at venturevoip.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
bilal ghayyad wrote:
| Hi List;
|
| With new technolgy, alot of mobiles now support Video
| Call, so what is the possibility to have Asterisk
| supporting Video so it support Video call at theie
| Phones?
Have a look at sip.fontventa.com as well as the Asterisk-Video mailing
list.
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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------------------------------
Message: 7
Date: Mon, 14 Jan 2008 18:47:52 -0600
From: "Anton Krall" <akrall at intruder.com.mx>
Subject: Re: [asterisk-users] app_voicemail for spanish
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<330417F4B6BCA34C917684152699E21405556A at mail.exchange.intruder.com.mx>
Content-Type: text/plain; charset="us-ascii"
Im looking at app_voicemail (remember, this is on 1.2.x) and there seems
to be some syntax changes for Spanish but doesn't seem to have all
that's required... Ill file a bug report on mantis.
AK
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew
Joakimsen
Sent: lunes, 14 de enero de 2008 05:58 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] app_voicemail for spanish
The language support is supposed to be there I know I've played with
it and there are at least SOME grammatical changes (don't recall which
right now)
But if further language support is needed you should file a bugreport.
On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote:
| Quote: | Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying "trabajo
|
mensjes"
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
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------------------------------
Message: 8
Date: Mon, 14 Jan 2008 19:50:22 -0500
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<ea18e54a0801141650o242877b7te9e68882b5d05237 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Jan 14, 2008 6:54 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
| Quote: | On Jan 14, 2008 5:51 PM, Steve Totaro <stotaro at totarotechnologies.com>
wrote:
| Quote: |
Either that or pay for the legal licensing of G729 and get support
|
through
| Quote: | the appropriate channels. Using the code for anything other than
|
learning
| Quote: | purposes is illegal, not to mention that licensing is quite
inexpensive.
|
Using the code period in a country which recognizes software patents
is an infringement of the patentholder rights. It is not illegal
anywhere but it does open you up to a great deal of legal liability.
It does not matter if its in production use or not it is still
infringement on the patent. Of course unless you have a large
operation, say the size of Vonage, noone's really going to care.. but
why are you going to start small with that sort of thinking? You'll
never get anywhere.
|
I would argue that it is illegal. The main definition of illegal is
"1. *against
law: *contravening a specific law, especially a criminal law".
http://encarta.msn.com/dictionary_/illegal.html
While it may not be against criminal law in the US it can be in France and
Austria, in the US it is certainly "against a specific law".
http://en.wikipedia.org/wiki/Patent_law#Law
Anyways, buying the license is the right thing to do unless you live where
software patent laws are not applicable.
| Quote: |
I wonder how many Chinese VoIP phones with G729 & G723 codecs have
actually licensed the codec?
|
Probably none.
Thanks,
Steve Totaro
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------------------------------
Message: 9
Date: Tue, 15 Jan 2008 06:22:27 +0530
From: "Mayur" <mninama at varaha.com>
Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF
conversion erros.
To: <david.cantera at IBSOneCall.com>, "'Asterisk Users Mailing List -
Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
Message-ID:
<mailman.10082.1200389991.10646.asterisk-users at lists.digium.com>
Content-Type: text/plain; charset="us-ascii"
Hi David,
Thank you for suggestion. It seems to work well. So asterisk does inband
dtmf to SIP INFO dtmf conversion well. I am curious to know why there is
no
consistency with 2833 to INFO DTMF conversion. Is it a known issue with
asterisk?
Regards,
Mayur
_____
From: dave cantera [mailto:david.cantera at iacnet.net]
Sent: Sunday, January 13, 2008 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
mninama at varaha.com
Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion
erros.
mayur,
did you try inband? with sip?
daveC
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;allow=ulaw ; dtmfmode=inband only works with ulaw or
alaw!
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;allow=ulaw ; dtmfmode=inband only works with ulaw or
alaw!
Mayur wrote:
Hi,
I am using asterisk 1.4.17 which is connected to a SIP trunk supporting
rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have
set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and
for
SIP clients I have set dtmfmode=info. So when I make a call to a cell
number
using the sip trunk and then press digits I can see the 2833 dtmf events
coming to asterisk in the rtp captures. Asterisk seems to detect those and
give SIP INFO to the SIP client. However it fails to detect some of the
digits (which is random) hence the correct sequence of digits is not
received at the SIP client.
I have tried setting relaxdtmf=yes in sip.conf but that does not seem to
help. Can anyone help me out here?
Regards,
Mayur
_____
_______________________________________________
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Message: 10
Date: Mon, 14 Jan 2008 16:58:13 -0800 (PST)
From: Steve Edwards <asterisk.org at sedwards.com>
Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP
script returned -1
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Pine.LNX.4.64.0801141647310.21507 at fs.sedwards.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Tue, 15 Jan 2008, Matt Riddell wrote:
| Quote: | Brian Hutchinson wrote:
|
| Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No
matter
| what my script returns (0 or -1), AGISTATUS always appears to be 0 =
| SUCCESS.
Why don't you just set a variable from the AGI and then test for it in
the dialplan
|
* The exit behavior of the AGI applications has changed. Previously, when
a connection to an AGI server failed, the application would cause the
channel
to immediately stop dialplan execution and hangup. Now, the only time
that
the AGI applications will cause the channel to stop dialplan execution
is
when the channel itself requests hangup. The AGI applications now set an
AGISTATUS variable which will allow you to find out whether running the
AGI
was successful or not.
Previously, there was no way to handle the case where Asterisk was
unable to
locally execute an AGI script for some reason. In this case, dialplan
execution will continue as it did before, but the AGISTATUS variable
will be
set to "FAILURE".
A locally executed AGI script can now exit with a non-zero exit code and
this
failure will be detected by Asterisk. If an AGI script exits with a
non-zero
exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
"SUCCESS".
I find the idea of a proliferation of inconsistently implemented AGI
failure or success variables undesirable. As I read the above, if
returning a non-zero exit code does not set AGISTATUS to "FAILURE," it's a
bug that needs to be reported.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
------------------------------
Message: 11
Date: Tue, 15 Jan 2008 03:06:46 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: asterisk-users at lists.digium.com
Message-ID: <20080115010646.GV32205 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Mon, Jan 14, 2008 at 07:50:22PM -0500, Steve Totaro wrote:
| Quote: | On Jan 14, 2008 6:54 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
| Quote: | I wonder how many Chinese VoIP phones with G729 & G723 codecs have
actually licensed the codec?
|
Probably none.
|
Well, they sell in the US and in other countries. I suspect that if
licensing requirements were not satisfied, their reselers would have to
pay the licensing fees instead.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 12
Date: Mon, 14 Jan 2008 21:02:59 -0800
From: Rob <asterisk at private.eklhq.com>
Subject: [asterisk-users] Park() help, extension not heard
To: asterisk-users at lists.digium.com
Message-ID: <478C3E83.9060100 at private.eklhq.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm trying to get call parking to work, but I've run out of things to try.
I can place a call between two internal extensions, then on one
extension transfer the call to extension 700, and the call gets parked
on 701 but I don't hear the extension number when I do the transfer. I
can hangup and call 701 and get the call back.
Here's what I see: (comments added on lines starting with !!)
!! Start call from desktop to phone
-- Executing [*00 at internal:1] Macro("SIP/rob_desktop-007fbcb0",
"ring-all") in new stack
-- Executing [s at macro-ring-all:1] Dial("SIP/rob_desktop-007fbcb0",
"SIP/gs100|20") in new stack
-- Called gs100
-- SIP/gs100-00816bf0 is ringing
!! Answer the call
-- SIP/gs100-00816bf0 answered SIP/rob_desktop-007fbcb0
!! Press "transfer" button on phone
-- Started music on hold, class 'default', on SIP/rob_desktop-007fbcb0
== Spawn extension (macro-ring-all, s, 1) exited non-zero on
'SIP/rob_desktop-007fbcb0'
!! Dial "700" and "send" on phone
-- Started music on hold, class 'default', on SIP/rob_desktop-007fbcb0
== Parked SIP/rob_desktop-007fbcb0 on 701 at parkedcalls. Will timeout
back to extension [macro-ring-all] s, 1 in 45 seconds
-- <SIP/gs100-00816bf0> Playing 'digits/7' (language 'en')
-- <SIP/gs100-00816bf0> Playing 'digits/0' (language 'en')
-- <SIP/gs100-00816bf0> Playing 'digits/1' (language 'en')
!! Hear "beep" on phone
-- Added extension '701' priority 1 to parkedcalls
-- Stopped music on hold on SIP/rob_desktop-007fbcb0
== SIP/rob_desktop-007fbcb0 got tired of being parked
It looks like it's doing the right thing, but I never hear "7" "0" "1".
I hear a "beep" after the "1" message is logged.
I added an extension that does Answer(), SayDigits(123), and Hangup(),
and I hear "one" "two" "three" perfectly.
What do I need to do to hear the extension where the call gets parked?
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Message: 13
Date: Tue, 15 Jan 2008 08:10:37 +0300
From: "Brian Hutchinson" <b.hutchman at gmail.com>
Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP
script returned -1
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Cc: matt at venturevoip.com
Message-ID:
<3d1967ab0801142110x4147fb64yda1cafe765a373d7 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
| Quote: |
Why don't you just set a variable from the AGI and then test for it in
|
That is what I ended up doing and that worked. Just thought I'd post to
the
list since from what I read it sounds like the script return value should
be
reflected in AGISTATUS and it wasn't. Didn't know if it was a bug that
should be reported or not.
Thanks for your help.
Regards,
Brian
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Message: 14
Date: Tue, 15 Jan 2008 08:16:04 +0300
From: "Brian Hutchinson" <b.hutchman at gmail.com>
Subject: Re: [asterisk-users] Asterisk 1.4.17 crashing more
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Cc: abdul_zu at yahoo.com
Message-ID:
<3d1967ab0801142116h52205ae0t3d08c5acdfd8e65e at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm running 1.4.17. I've been running that version plus an addition of
Unicall MFC/R2 and the only time I have seen it die is right away on
startup
due to something in one of the .conf files not being right. It has not
died
during normal operation. I'm running two TE420B cards on a large Dell
2950. Not doing SIP so I'm not exercising that portion of the code.
Regards,
Brian
On Jan 15, 2008 2:23 AM, Abdul <abdul_zu at yahoo.com> wrote:
| Quote: | Hi All,
We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one
day it stop to response to the SIP Clinets so they cannot make call or
register. But safe_asterisk not restarting it back because asterisk
running
without any response to the sip clients.
When we try to do 'core show channels' using Manager it returns only
Action: Command
Command: show channels
That time asterisk not responding anything for clients for registration
either for invitation.
Please advice us how we can fix this issue.
------------------------------
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Message: 15
Date: Tue, 15 Jan 2008 00:48:11 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] app_voicemail for spanish
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801142148i1ee66cc6o64411b43de0c7a4f at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
No features are being added for 1.2 so I'd check to see if 1.4 has the
changes you need before filing a bugreport.
On Jan 14, 2008 7:47 PM, Anton Krall <akrall at intruder.com.mx> wrote:
| Quote: | Im looking at app_voicemail (remember, this is on 1.2.x) and there seems
to be some syntax changes for Spanish but doesn't seem to have all
that's required... Ill file a bug report on mantis.
AK
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew
Joakimsen
Sent: lunes, 14 de enero de 2008 05:58 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] app_voicemail for spanish
The language support is supposed to be there I know I've played with
it and there are at least SOME grammatical changes (don't recall which
right now)
But if further language support is needed you should file a bugreport.
On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote:
| Quote: | Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying "trabajo
|
mensjes"
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------------------------------
Message: 16
Date: Tue, 15 Jan 2008 00:57:13 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801142157p69ab6be2i53b38a9dfdc8eb3d at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
On Jan 14, 2008 7:50 PM, Steve Totaro <stotaro at totarotechnologies.com>
wrote:
Illegal means that something violates a criminal law. You linked to a
page that describe the law in the US regarding patentholders
registration of said patents. I'm not saying we should infringe on the
patentholder's right I am simply saying it is not a criminal act, at
least in the US.
Software is generally not patentable in the European Union (and
probably in the countries that are pseudo-EU members)
| Quote: | Anyways, buying the license is the right thing to do unless you live
where
software patent laws are not applicable.
|
Totally agree.
------------------------------
Message: 17
Date: Mon, 14 Jan 2008 22:34:05 -0800
From: Rob <asterisk at private.eklhq.com>
Subject: Re: [asterisk-users] Park() help, extension not heard
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478C53DD.6080008 at private.eklhq.com>
Content-Type: text/plain; charset=ISO-8859-1
1.4.17.
Rob wrote:
| Quote: | I'm trying to get call parking to work, but I've run out of things to
try.
I can place a call between two internal extensions, then on one
extension transfer the call to extension 700, and the call gets parked
on 701 but I don't hear the extension number when I do the transfer.
I can hangup and call 701 and get the call back.
Here's what I see: (comments added on lines starting with !!)
!! Start call from desktop to phone
-- Executing [*00 at internal:1] Macro("SIP/rob_desktop-007fbcb0",
"ring-all") in new stack
-- Executing [s at macro-ring-all:1] Dial("SIP/rob_desktop-007fbcb0",
"SIP/gs100|20") in new stack
-- Called gs100
-- SIP/gs100-00816bf0 is ringing
!! Answer the call
-- SIP/gs100-00816bf0 answered SIP/rob_desktop-007fbcb0
!! Press "transfer" button on phone
-- Started music on hold, class 'default', on
SIP/rob_desktop-007fbcb0
== Spawn extension (macro-ring-all, s, 1) exited non-zero on
'SIP/rob_desktop-007fbcb0'
!! Dial "700" and "send" on phone
-- Started music on hold, class 'default', on
SIP/rob_desktop-007fbcb0
== Parked SIP/rob_desktop-007fbcb0 on 701 at parkedcalls. Will timeout
back to extension [macro-ring-all] s, 1 in 45 seconds
-- <SIP/gs100-00816bf0> Playing 'digits/7' (language 'en')
-- <SIP/gs100-00816bf0> Playing 'digits/0' (language 'en')
-- <SIP/gs100-00816bf0> Playing 'digits/1' (language 'en')
!! Hear "beep" on phone
-- Added extension '701' priority 1 to parkedcalls
-- Stopped music on hold on SIP/rob_desktop-007fbcb0
== SIP/rob_desktop-007fbcb0 got tired of being parked
It looks like it's doing the right thing, but I never hear "7" "0"
"1". I hear a "beep" after the "1" message is logged.
I added an extension that does Answer(), SayDigits(123), and Hangup(),
and I hear "one" "two" "three" perfectly.
What do I need to do to hear the extension where the call gets parked?
|
------------------------------
Message: 18
Date: Tue, 15 Jan 2008 08:17:19 +0100
From: Stefan Guenther <asterisk01 at in-put.de>
Subject: [asterisk-users] pickupchan without bristuffed version?
To: asterisk-users at lists.digium.com
Message-ID: <478C5DFF.7090008 at in-put.de>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Hello,
following the description in the wiki
(http://www.voip-info.org/wiki/view/Asterisk+phone+snom)
I have set up a number of SNOM phones to monitor extensions with hints.
The lights on the phones flash when a call on another phone comes in.
According to the article in the wiki I need the application PickUpChan
to catch on of the calls which causes the light to flash. But PickUpchan
is only available in the bristuff version of asterisk
Is there another way to get a specific call and not just press *8 to get
a random call out of the callgroup?
Thanks for your help,
Stefan
--
********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
********************************************
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
********************************************
------------------------------
Message: 19
Date: Tue, 15 Jan 2008 09:01:25 +0000
From: Bruce McAlister <bruce.mcalister at blueface.ie>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478C7665.1000404 at blueface.ie>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Steve Totaro wrote:
| Quote: |
I would suggest building it yourself
(http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt
<http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt>). It is
not that difficult and ensures that it "should" be compatible with your
machine. Just a little work.
|
Has anyone tried building this on Solaris, I just had a look at the link
and it looks like the Intel IPP stuff is only released for Windows,
Linux and MAC. And the v32 G729 codec from Digium does not load within
asterisk on Solaris, sooo, the Solaris users out there dont have much
support when it comes to G729 codecs, a real pity really, this stops
some large scale roll-outs.
------------------------------
Message: 20
Date: Tue, 15 Jan 2008 09:05:35 +0000
From: Thomas Kenyon <digium at sanguinarius.co.uk>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478C775F.4010002 at sanguinarius.co.uk>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Andrew Joakimsen wrote:
| Quote: | On Jan 14, 2008 7:50 PM, Steve Totaro <stotaro at totarotechnologies.com>
wrote:
| Quote: | Anyways, buying the license is the right thing to do unless you live
where
software patent laws are not applicable.
|
Totally agree.
|
I have bought many more licenses from asterisk than I've ever used, and
mostly use the asterisk.hosting.lv codecs.
Twice now while using the digium codec, upon upgrading asterisk, it
stopped working.
The Beta codec (based on IPP5), is much much faster than either the
digium or the older codec, and at home (only place I run beta software),
there hasn't been a problem.
Mind you, according to show translation, the older codec (based on
IPP4), is faster than the digium codec too.
------------------------------
Message: 21
Date: Tue, 15 Jan 2008 04:23:04 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: bruce.mcalister at blueface.ie, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801150123u66de469dm14fd4b31cf0fddfa at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
They used to have solaris on the Digium FTP site but they seem to be gone
now 
On the "free" codec site they have some complied with icc and others
with gcc4 so I don't see why you can't get this working with gcc on
solaris.
On Jan 15, 2008 4:01 AM, Bruce McAlister <bruce.mcalister at blueface.ie>
wrote:
| Quote: | Steve Totaro wrote:
| Quote: |
I would suggest building it yourself
(http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt
<http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt>). It
is
not that difficult and ensures that it "should" be compatible with your
machine. Just a little work.
|
Has anyone tried building this on Solaris, I just had a look at the link
and it looks like the Intel IPP stuff is only released for Windows,
Linux and MAC. And the v32 G729 codec from Digium does not load within
asterisk on Solaris, sooo, the Solaris users out there dont have much
support when it comes to G729 codecs, a real pity really, this stops
some large scale roll-outs.
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------------------------------
Message: 22
Date: Tue, 15 Jan 2008 09:39:16 +0000
From: Thomas Kenyon <digium at sanguinarius.co.uk>
Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk
1.2.x.
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478C7F44.1020406 at sanguinarius.co.uk>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Andrew Joakimsen wrote:
| Quote: | They used to have solaris on the Digium FTP site but they seem to be gone
now
On the "free" codec site they have some complied with icc and others
with gcc4 so I don't see why you can't get this working with gcc on
solaris.
|
If you can, be sure to submit it to arkadi at kvin.lv , I'm sure he'll be
happy to receive it.
------------------------------
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