[asterisk-users] asterisk-users Digest, Vol 42, Issue 51
sandeep
sandeep.s at briotelecom.com
Wed Jan 16 23:33:03 CST 2008
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
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> 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:
>> 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
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> ------------------------------
>
> 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:
>
>> 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)
> -----BEGIN PGP SIGNATURE-----
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>
> iD8DBQFHi/ykDQNt8rg0Kp4RAh93AJ9BOV/+IjAqte/coiOTCAciRzI25wCZAYW3
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> =4oUM
> -----END PGP SIGNATURE-----
>
>
>
> ------------------------------
>
> 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)
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFHi/29DQNt8rg0Kp4RAmNTAJwNylh0kXzAVeRKXfrkmi9KPjeTrQCeISQN
> ognhVpn4zXNu+QR+Rp3quPA=
> =yhjW
> -----END PGP SIGNATURE-----
>
>
>
> ------------------------------
>
> 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:
>> 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
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> 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:
>
>> On Jan 14, 2008 5:51 PM, Steve Totaro <stotaro at totarotechnologies.com>
>> wrote:
>> >
>> > Either that or pay for the legal licensing of G729 and get support
>> through
>> > the appropriate channels. Using the code for anything other than
>> learning
>> > 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.
>
>
>
>>
>> I wonder how many Chinese VoIP phones with G729 & G723 codecs have
>> actually licensed the codec?
>>
>>
> Probably none.
>
>
> Thanks,
> Steve Totaro
> -------------- next part --------------
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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
>
>
>
>
>
>
>
> _____
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
>
> _____
>
>
>
>
> Internal Virus Database is out-of-date.
> Checked by AVG Free Edition.
> Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date:
> 01/04/2008
> 12:05 PM
>
>
>
>
>
>
> --
> My wife's sister is in California.
> I should buy her a Videophone2008!
>
> Truly, The Next Best Thing to Being There!
> --
>
> WorldWideVideoPhones.com
> 856.380.0894
>
>
>
>
>
> __________ NOD32 2786 (20080112) Information __________
>
> This message was checked by NOD32 antivirus system.
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>
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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:
>
>> 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
>
>>From UPGRADE.txt:
>
> * 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:
>> On Jan 14, 2008 6:54 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
>> > 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"
>
>>
>>
>>
>> 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
>
>
>> 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)
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.7 (MingW32)
>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>>
>> iD8DBQFHi/ykDQNt8rg0Kp4RAh93AJ9BOV/+IjAqte/coiOTCAciRzI25wCZAYW3
>> BW/ubpchpy2KUQROsmPnonQ=
>> =4oUM
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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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:
>
>> 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.
>>
>>
>> ------------------------------
>> Looking for last minute shopping deals? Find them fast with Yahoo!
>> Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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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:
>> 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:
>> > 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
>> >
>> >
>> > _______________________________________________
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> ------------------------------
>
> 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:
>>
>>
>
>> 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
>
> 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.
>
>> 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
>
> Software is generally not patentable in the European Union (and
> probably in the countries that are pseudo-EU members)
>
>> 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:
>> 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:
>
>>
>> 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:
>> On Jan 14, 2008 7:50 PM, Steve Totaro <stotaro at totarotechnologies.com>
>> wrote:
>>>
>>
>>> 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:
>> Steve Totaro wrote:
>>
>> >
>> > 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.
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> ------------------------------
>
> 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:
>> 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.
>
>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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