[Asterisk-Dev] Re: Asterisk-Dev digest, Vol 1 #597 - 9 msgs

[Dan]

On Fri, 23 Apr 2004 12:00:01 -0500, asterisk-dev-request at lists.digium.com wrote:
> Send Asterisk-Dev mailing list submissions to
> asterisk-dev at lists.digium.com
>
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-dev or, via
> email, send a message with subject or body 'help' to asterisk-dev-
> request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-dev-admin at lists.digium.com
>
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-Dev digest..."
>
>
> Today's Topics:
>
>
> 1. Problem with DND working (kevin at mail.rnktel.com) 2. Re:
> [Asterisk-cvs] zaptel patgen.c,1.2,1.3 pattest.c,1.2,1.3
> wct4xxp.c,1.40,1.41 (Tais M. Hansen) 3. G.729 Licenses for Asterisk
> (Sudhir Kumar)
> 4. Re: G.729 Licenses for Asterisk (Rich Adamson) 5. Re: Re:
> [Asterisk-cvs] zaptel patgen.c,1.2,1.3 pattest.c,1.2,1.3
> wct4xxp.c,1.40,1.41 (reseaux)
> 6. Re: G.729 Licenses for Asterisk (Eric Wieling) 7. Re: Problem
> with DND working (Steven Critchfield) 8. Re: Transfer of variables
> in app_dial (Mike Sturdee) 9. RE: Problem with DND working (brian)
>
> --__--__--
>
Kevin what don`t you just use something like 

[macro-foraccounts]
exten => s,1,DBget(temp=SIP/${ARG1})
exten => s,2,Goto(from-sip,${temp}|1)
exten => s,102,Goto(s|3)
exten => s,3,DBget(dnd=DND/${ARG1})
exten => s,4,Goto(s|6)
exten => s,104,Goto(s|5)
exten => s,5,Dial(SIP/${ARG1},30,Ttri)
exten => s,6,Voicemail2(b${ARG1})
exten => s,7,Hangup
exten => s,106,Voicemail2(b${ARG1}) ; busy
exten => s,107,Hangup


[dnd] ; DO NOT DISTURB
exten => _*61,1,DBput(DND/${CALLERIDNUM}=YES})
exten => _*61,2,Playback(vm-goodbye)
exten => _*61,3,SoftHangup
exten => _*60,1,DBdel(DND/${CALLERIDNUM})
exten => _*60,2,Playback(vm-goodbye)
exten => _*60,3,SoftHangup

Because with this DND works great
>
> Message: 1
> Date: Fri, 23 Apr 2004 08:49:09 -0400 (EDT)
> From: kevin at mail.rnktel.com
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] Problem with DND working
> Reply-To: asterisk-dev at lists.digium.com
>
>
> Hey
>
>
> I have been trying to get DND working using 3 separate AGI scripts
> with no success.
>
> dndon.agi - Writes to the database and if you do a database show
> you see /DND/exten#
>
>
> dndoff.agi - Does the database del and removes the entry.
>
>
> the problem is with my dndck.agi. This is the checker to see if the
> party has DND turned on. No matter what I try to do a match on it
> fails. Here is the copy of the AGI I'm using:
>
> #! /usr/bin/perl -w
>
>
> use Asterisk::AGI;
>
>
> my $AGI = new Asterisk::AGI;
>
>
> my $var = $AGI->database_get(DND,1000);
>
>
> if($var eq 'YES') {
>
>
> $AGI->exec(Voicemail','u1000');
>
>
> }else{
>
>
> $AGI->exec('Dial','Sip/username');
>
>
> }
>
>
> Any help would be greatly appreciated.
>
>
> Thanks
> Kevin
>
>
> --__--__--
>
>
> Message: 2
> From: "Tais M. Hansen" <tmh at comx.as>
> Organization: ComX Networks
> To: asterisk-dev at lists.digium.com
> Date: Fri, 23 Apr 2004 15:02:37 +0200
> Subject: [Asterisk-Dev] Re: [Asterisk-cvs] zaptel patgen.c,1.2,1.3
> pattest.c,1.2,1.3 wct4xxp.c,1.40,1.41
> Reply-To: asterisk-dev at lists.digium.com
>
>
> =2D----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>
> On Friday 23 April 2004 05:05, jim at lists.digium.com wrote:
>
>> Modified Files:
>> patgen.c pattest.c wct4xxp.c
>> Log Message:
>> Fixed problems in wct4xxp driver, and added error/alarm
>> processing to patgen and pattest (Jim D)
>>
>
> These changes locks up my system. It seems to be working fine if I
> grab zap= tel=20 from yesterday.
>
> =2D --=20
> Regards,
> Tais M. Hansen
> ComX Networks
> Tel: +45-70257474
> =46ax: +45-70257374
>
>
> =2D----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.2.3 (GNU/Linux)
>
>
> iD8DBQFAiRPz2TEAILET3McRAuzJAJ9DuDM+g2Mqwlxw1s7uxdkEtp4v7QCeOHe7
> WMnSZAkYoclRY7ElvE59Uhc=3D =3D0mBc =2D----END PGP SIGNATURE-----
>
> --__--__--
>
>
> Message: 3
> From: Sudhir Kumar <sudhir1 at adelphia.net>
> To: asterisk-dev at lists.digium.com
> Organization:
> Date: 23 Apr 2004 09:13:42 -0400
> Subject: [Asterisk-Dev] G.729 Licenses for Asterisk Reply-To:
> asterisk-dev at lists.digium.com
>
> We are preparing an Asterisk PBX for a client who is going to use
> this with Grandstream phones. All his external calls are routed
> through another provider's Cisco AS5300 box.
>
> Now my question is how many G.729 license's does he need to
> purchase? Both Grandstream phones and AS5300 do G.729, hence all
> the calls are in pass through mode as far as G.729 is concerned. It
> is unlikely that more than 5 people will be checking their voice
> mails. Should 5 G.729 licenses be sufficient, or need more?
>
> Although it does not matter much in this case as extra $100 for 10
> more licenses is not a big money for this client yet I thought I
> will seek clarification as this issue will come up again and again.
>
> Thanks,
> -- sudhir
>
>
> --__--__--
>
>
> Message: 4
> Date: Fri, 23 Apr 2004 08:27:31 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Dev] G.729 Licenses for Asterisk To:
> asterisk-dev at lists.digium.com Reply-To: asterisk-
> dev at lists.digium.com
>
>> We are preparing an Asterisk PBX for a client who is going to use
>> this with Grandstream phones. All his external calls are routed
>> through another provider's Cisco AS5300 box.
>>
>> Now my question is how many G.729 license's does he need to
>> purchase? Both Grandstream phones and AS5300 do G.729, hence all
>> the calls are in pass through mode as far as G.729 is concerned.
>> It is unlikely that more than 5 people will be checking their
>> voice mails. Should 5 G.729 licenses be sufficient, or need more?
>>
>> Although it does not matter much in this case as extra $100 for
>> 10 more licenses is not a big money for this client yet I thought
>> I will seek clarification as this issue will come up again and
>> again.
>>
>
> I can't answer your question directly (and yes I'm running a
> limited number of licenses as well), however it seems previous
> posters have not taken into consideration some of the incidental
> license needs. Not sure it any of these fit, however think about: -
> voicemail (gsm or whatever to 729)
> - access to pstn facilities that do not support 729 - other *
> services that you might have in your dialplan, etc, whose sounds
> are derived from non-729 sources
> - consultive transfers, meetme, and other such services
>
>
> Better to error on the too many side.
>
>
> --__--__--
>
>
> Message: 5
> From: reseaux <reseauxit at yahoo.it>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Re: [Asterisk-cvs] zaptel
> patgen.c,1.2,1.3 pattest.c,1.2,1.3 wct4xxp.c,1.40,1.41
> Date: Fri, 23 Apr 2004 15:43:56 +0200
> Reply-To: asterisk-dev at lists.digium.com
>
>
> Dear
> I have update the zaptel from cvs now and i have the same issue
> with my * box and  TE410P with this new CVS version i have must
> turn back to yestarday..
>
> On Friday 23 April 2004 03:02 pm, Tais M. Hansen wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>>
>> On Friday 23 April 2004 05:05, jim at lists.digium.com wrote:
>>
>>> Modified Files:
>>> patgen.c pattest.c wct4xxp.c
>>> Log Message:
>>> Fixed problems in wct4xxp driver, and added error/alarm
>>> processing to patgen and pattest (Jim D)
>>>
>>
>> These changes locks up my system. It seems to be working fine if
>> I grab zaptel from yesterday.
>>
>> - --
>> Regards,
>> Tais M. Hansen
>> ComX Networks
>> Tel: +45-70257474
>> Fax: +45-70257374
>>
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.2.3 (GNU/Linux)
>>
>>
>> iD8DBQFAiRPz2TEAILET3McRAuzJAJ9DuDM+g2Mqwlxw1s7uxdkEtp4v7QCeOHe7
>> WMnSZAkYoclRY7ElvE59Uhc= =0mBc -----END PGP SIGNATURE-----
>> _______________________________________________ Asterisk-Dev
>> mailing list Asterisk-Dev at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-dev To
>> UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>
> --__--__--
>
>
> Message: 6
> Subject: Re: [Asterisk-Dev] G.729 Licenses for Asterisk From: Eric
> Wieling <eric at fnords.org> To: asterisk-dev at lists.digium.com
> Organization: BTEL Consulting Date: Fri, 23 Apr 2004 09:02:36 -0500
> Reply-To: asterisk-dev at lists.digium.com
>
>>> Now my question is how many G.729 license's does he need to
>>> purchase? Both Grandstream phones and AS5300 do G.729, hence
>>> all the calls are in pass through mode as far as G.729 is
>>> concerned. It is unlikely that more than 5 people will be
>>> checking their voice mails. Should 5 G.729 licenses be
>>> sufficient, or need more?
>>>
>
> My general policy is to buy twice as many licenses as I think I'll
> need.  The licenses are cheap and I've been told that adding
> licenses can be somewhat of a hassle.
>
> --
> Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related
> story, the IRS has recently ruled that the cost of Windows upgrades
> can NOT be deducted as a gambling loss."
>
>
> --__--__--
>
>
> Message: 7
> Subject: Re: [Asterisk-Dev] Problem with DND working From: Steven
> Critchfield <critch at basesys.com> To: asterisk-dev at lists.digium.com
> Date: Fri, 23 Apr 2004 08:43:52 -0500 Reply-To: asterisk-
> dev at lists.digium.com
>
> On Fri, 2004-04-23 at 07:49, kevin at mail.rnktel.com wrote:
>
>> #! /usr/bin/perl -w
>> use Asterisk::AGI;
>> my $AGI = new Asterisk::AGI;
>> my $var = $AGI->database_get(DND,1000);
>>
>
> Can't help for the above section, but instead of execing things
> below, you should just upgrade the priority to th proper amount so
> when you exit the dial plan handles the call.
>
>> if($var eq 'YES') {
>> $AGI->exec(Voicemail','u1000');
>> }else{
>> $AGI->exec('Dial','Sip/username');
>> }
> --
> Steven Critchfield <critch at basesys.com>
>
>
> --__--__--
>
>
> Message: 8
> Date: Fri, 23 Apr 2004 10:13:04 -0400 (EDT)
> From: Mike Sturdee <sturdee at pathwaynet.com>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Transfer of variables in app_dial Reply-
> To: asterisk-dev at lists.digium.com
>
> On Tue, 20 Apr 2004, Olle E. Johansson wrote:
>
>
>> John Todd wrote:
>>> At 6:15 PM +0200 on 4/20/04, Olle E. Johansson wrote:
>>>
>>>
>>>> In some cases, you want channel variables to be transferred
>>>> from the caller to the callee.
>>>> This works for VXML_URL but not for other channels.
>>>>
>>>>
>>>> There's a patch that transfers all variables in
>>>> http://bugs.digium.com/bug_view_page.php?bug_id=0000928
>>>>
>>>> I discussed this with Mark and we agreed upon a solution that
>>>> we want feedback on:
>>>>
>>>> * Not transfer all variables by default (as in the current
>>>> patch) * Transfer all variables that begin with "_"
>>>> (underscore) once and remove the underscore in the new
>>>> channel leg * Transfer all variables that begin with "__"
>>>> (two underscore) without removing anything, making it stay
>>>> even if the new leg is transferred to a third leg
>>>>
>>
>>> Hmmm... sounds like a very reasonable plan.  There are two
>>> concepts here, of course: the transfer of values within
>>> Asterisk during a Local transfer (which I think is already
>>> handled by using or not using the /n option in Dial, IIRC) but
>>> then there is the more significant and vexing problem of
>>> transferring values OUT of a particular server to a remote
>>> server.
>>>
>> Well, this patch takes care of local transfers. To explain to
>> other readers:
>>
>>
>> When you exec dial() in the dial plan, you're in one call leg.
>> Dial() opens another call leg to attach your call to some other
>> device. We now have two call legs.
>>
>> The channel variables you set with setvar is local to the current
>> call leg, the caller. Sometimes we want variables to be
>> transferred to the other leg, the callee. This patch would handle
>> that.
>>
>> John brings another topic up for discussion, actually
>> transferring variable names and values over the link out of band
>> to the other end. This would be very useful when clustering
>> asterisk servers, transferring account codes and other variables.
>>
>> I need a solution like this to handle SIP 302 transfers, to be
>> able to send a transfer back to the dial plan again.
>>
>> /Olle
>>
>
> One other thing that would be useful is the passing of variables
> from a caller in a Queue to the agent. At this point, not even the
> Queue name itself can be found in a variable, and a uniqueid that
> follows a call from Incoming -> Zap -> AA -> Queue -> Agent ->
> Hangup would be very useful for CDR purposes.
>
> -Mike
>
>
> ==================================
> Network Engineer
> Pathway Internet Services
> 616.774.3131
>
>
> --__--__--
>
>
> Message: 9
> From: "brian" <brian at bkw.org>
> To: <asterisk-dev at lists.digium.com>
> Subject: RE: [Asterisk-Dev] Problem with DND working Date: Fri, 23
> Apr 2004 09:27:44 -0500 Organization: BKW.ORG Reply-To: asterisk-
> dev at lists.digium.com
>
> Why even use agi.. this can be done in the dial plan with a lot
> less drama... app_db includes dbget, dbput, dbdel and dbdeltree
>
> bkw
>
>
>> -----Original Message-----
>> From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
>> admin at lists.digium.com] On Behalf Of kevin at mail.rnktel.com Sent:
>> Friday, April 23, 2004 7:49 AM To: asterisk-dev at lists.digium.com
>> Subject: [Asterisk-Dev] Problem with DND working
>>
>>
>> Hey
>>
>>
>> I have been trying to get DND working using 3 separate AGI
>> scripts with no success.
>>
>> dndon.agi - Writes to the database and if you do a database show
>> you see /DND/exten#
>>
>>
>> dndoff.agi - Does the database del and removes the entry.
>>
>>
>> the problem is with my dndck.agi. This is the checker to see if
>> the party has DND turned on. No matter what I try to do a match
>> on it fails. Here is the copy of the AGI I'm using:
>>
>> #! /usr/bin/perl -w
>>
>>
>> use Asterisk::AGI;
>>
>>
>> my $AGI = new Asterisk::AGI;
>>
>>
>> my $var = $AGI->database_get(DND,1000);
>>
>>
>> if($var eq 'YES') {
>>
>>
>> $AGI->exec(Voicemail','u1000');
>>
>>
>> }else{
>>
>>
>> $AGI->exec('Dial','Sip/username');
>>
>>
>> }
>>
>>
>> Any help would be greatly appreciated.
>>
>>
>> Thanks
>> Kevin
>> _______________________________________________
>> Asterisk-Dev mailing list
>> Asterisk-Dev at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-dev To
>> UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> --__--__--
>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> End of Asterisk-Dev Digest