[Asterisk-Users] Looking for PRI Outbound Caller ID Configuration

2005-06-20 Thread Shaun Tierney
I'm having trouble setting the outbound caller ID on calls I make from my
PRI trunk group.  The PRIs are served out of a 5ESS.  Telco has set the PRIs
up for user provided caller id information, so I believe I just don't have
it set up right in my dialplan or something.  I can't seem to find an
example of setting the outbound caller ID specifically for a 5ESS.  Does
anyone have an example configuration that they have used with a 5ESS switch?
Below is the my configuration from Zapata.conf and a sample extension I've
tried to use to connect a call with new caller ID information provided by my
PBX.  Any insight is most appreciated.

[channels]
priindication = outofband
usecallerid=yes
cidsignalling=bell
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
switchtype=5ess
context=main
signalling=pri_cpe
group=1
channel = 1-23
channel = 25-47

exten = 1234,1,Wait,1
exten = 1234,2,Answer
exten = 1234,3,SetCallerPres(allowed_passed_screen)
exten = 1234,4,SetCIDNum(8881234567)
exten = 1234,5,Dial(Zap/g1/18887654321,,,)
exten = 1234,6,Hangup

Thanks,

Shaun Tierney

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RE: [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration

2005-06-20 Thread Shaun Tierney
That didn't seem to work either.  Any other ideas?

Thanks,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, June 20, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Looking for PRI Outbound Caller ID
Configuration

 I'm having trouble setting the outbound caller ID on calls I make from my
 PRI trunk group.  The PRIs are served out of a 5ESS.  Telco has set the
PRIs
 up for user provided caller id information, so I believe I just don't have
 it set up right in my dialplan or something.  I can't seem to find an
 example of setting the outbound caller ID specifically for a 5ESS.  Does
 anyone have an example configuration that they have used with a 5ESS
switch?
 Below is the my configuration from Zapata.conf and a sample extension I've
 tried to use to connect a call with new caller ID information provided by
my
 PBX.  Any insight is most appreciated.
 
 [channels]
 priindication = outofband
 usecallerid=yes
 cidsignalling=bell
 hidecallerid=no
 callwaiting=no
 usecallingpres=yes
 callwaitingcallerid=no
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=no
 rxgain=0.0
 txgain=0.0
 group=1
 callgroup=1
 pickupgroup=1
 immediate=no
 switchtype=5ess
 context=main
 signalling=pri_cpe
 group=1
 channel = 1-23
 channel = 25-47
 
 exten = 1234,1,Wait,1
 exten = 1234,2,Answer
 exten = 1234,3,SetCallerPres(allowed_passed_screen)
 exten = 1234,4,SetCIDNum(8881234567)
 exten = 1234,5,Dial(Zap/g1/18887654321,,,)
 exten = 1234,6,Hangup
 

Try something like this...
exten = _1NX,1,SetCallerID(8881234567|a)
exten = _1NX,2,SetCIDName(MyName|a)
exten = _1NX,3,Dial(ZAP/g1/${EXTEN})


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[Asterisk-Users] RE: LOA for CFA . . work up pencil copy

2005-06-06 Thread Shaun Tierney
David,

I guess I'm a little confused here.  Are you asking me to provide a pencil
copy of an LOA for your review?  I don't understand why you need an LOA
from us.  We need an LOA from you to order circuits that will be billed to
us that will be attached to your CFA.  It was also my understanding that you
had an LOA ready to be given to us, which had already been reviewed by your
lawyers.  I have been hearing about this LOA that SC has had in its
possession for probably at least a month now and I have asked about it on
numerous occasions yet it doesn't seem to be moving anywhere.  In the
conference call last week, my understanding was that it just needed to be
signed by an authorized SC Telcom employee like Steve and then sent to us.
Please take care of this ASAP as we need to get the ball rolling on ordering
interoffice transport circuits.  Not only do we need these for Indy,
Coffeyville, and our ATM DS3 from ASI, but we also need them for Augusta and
Andover to order DS3s back to Broadway for the collocation equipment moves.

Regards,

Shaun

-Original Message-
From: David Henning [mailto:[EMAIL PROTECTED] 
Sent: Monday, June 06, 2005 9:47 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; Eric Ryker; Zack
Odell
Subject: Re: LOA for CFA . . work up pencil copy

Shaun,

Go ahead and complete for Steve Davis (SCTelcom General Manager) to sign.
If you want me  Zack to review a pencil copy, that's fine.

When we all discussed this on 5/31/05 the concensus was that SCTelcom would
sign necessary LOA's for PrairieStream to get access to co-lo's that are
still in SCTelcom name, but all new circuits would be ordered under
PrairieStream and billed directly to PrairieStream.

* * P.S.* *   Sorry I couldn't make the 10 am call this morning as we're
still scrambling to finalize BillStream software conversion from CostGuard.

Thanks,

David Henning
SCTelcom
[EMAIL PROTECTED] 


 Shaun Tierney [EMAIL PROTECTED] 06/06/05 10:29AM 
David,

I'm just wondering where we are at on that CFA LOA.  I brought it up in our
conference this morning with Zack and Eric, but they did not know what the
status is.

Regards,

Shaun Tierney
Network Engineer
Prairie Stream Communications
200 Arco Pl, Suite 11
Independence, KS 67301
Toll-Free: (866) 331-9001 EXT 212
Phone: (620) 331-9000 EXT 212
Fax: (620) 331-1181
E-mail: [EMAIL PROTECTED] 
http://www.prairiestream.com/ 



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[Asterisk-Users] Random SIP Phone Problem

2005-04-18 Thread Shaun Tierney
I am currently running CVS-HEAD-04/15/05-13:15:00 and I have an issue that
just recently cropped up.  I upgraded to this version of Asterisk last
Friday and now twice in the last two hours, all of my Aastra SIP phones lose
service suddenly.  Network connectivity is still there between the phones
and the PBX, and I have restart Asterisk to fix the issue.  Would it be
worth my time to move to the latest CVS Asterisk release even though it has
only been three days since I installed the version in operation?  Or would I
be better off going with a previous CVS release to fix the problem?  I can't
use the stable release because I use macro arguments in the dial command.
Here are the error messages that seem to show up for the duration of the
problem.

Apr 18 13:43:50 NOTICE[16997] chan_sip.c: Peer 'brettb' is now UNREACHABLE!
Last qualify: 1045
Apr 18 13:44:03 VERBOSE[16997] logger.c: Don't know what to do if second
ROSE component is of type 0x6
Apr 18 13:44:07 NOTICE[16997] app_queue.c: Added interface 'SIP/brettb' to
queue 'psc'
Apr 18 13:44:14 NOTICE[16997] chan_sip.c: Peer 'brettb' is now REACHABLE!
(76ms / 2000ms)

Regards,

Shaun

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RE: [Asterisk-Users] Dial Macro Arguments

2005-04-15 Thread Shaun Tierney
Well, for what it's worth, I hope that some kind of argument feature will be
implemented in stable then.  Macros are extremely useful, especially with
arguments.

Regards,

Shaun Tierney

-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 14, 2005 6:03 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial Macro Arguments


the feature you are talking about is still not commited to stable. at
the moment it is only availabe in CVS HEAD. You can try to download
the patch and apply it, however I did not succeed in applying it to
1.0.7 so I had to use HEAD.

On 4/14/05, Shaun Tierney [EMAIL PROTECTED] wrote:
 Hello all!  I posted a message a while back about a problem I was having
 in December.  I was unable to send arguments to the macro in the dial
 command.  I was told back then to use ^ as the delimiter between the macro
 name and the arguments and that I had to upgrade to a newer version of
 Asterisk.  Now it appears that this does not work now that I have upgraded
 to Asterisk 1.0.7.  Was this feature removed or replaced?  Below is the
 error message I am receiving and the link to my original message.

 Apr 14 17:30:53 WARNING[3992]: app_macro.c:90 macro_exec: No such context
 'macro-getstartseconds^Zap/3-1' for macro 'getstartseconds^Zap/3-1'

 http://lists.digium.com/pipermail/asterisk-users/2004-December/075928.html

 Thanks,

 Shaun
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[Asterisk-Users] Dial Macro Arguments

2005-04-14 Thread Shaun Tierney
Hello all!  I posted a message a while back about a problem I was having
in December.  I was unable to send arguments to the macro in the dial
command.  I was told back then to use ^ as the delimiter between the macro
name and the arguments and that I had to upgrade to a newer version of
Asterisk.  Now it appears that this does not work now that I have upgraded
to Asterisk 1.0.7.  Was this feature removed or replaced?  Below is the
error message I am receiving and the link to my original message.

Apr 14 17:30:53 WARNING[3992]: app_macro.c:90 macro_exec: No such context
'macro-getstartseconds^Zap/3-1' for macro 'getstartseconds^Zap/3-1'

http://lists.digium.com/pipermail/asterisk-users/2004-December/075928.html

Thanks,

Shaun
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RE: [Asterisk-Users] Problem compiling asterisk-addons

2005-03-23 Thread Shaun Tierney
I believe that it looks in ../asterisk/ for the asterisk.h file.  If your
source directories for both Asterisk and the Asterisk Addons are within the
same directory, try renaming the asterisk-x.x.x directory to just asterisk
then recompile.  It should work for you after that.

Regards,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric
Sent: Wednesday, March 23, 2005 1:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem compiling asterisk-addons


Hi,

I am getting an error trying to compile the asterisk addons:

cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
make: *** [cdr_addon_mysql.o] Error 1

Can anyone suggest something I could try?

Thanks.
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[Asterisk-Users] Queue Question

2005-02-21 Thread Shaun Tierney
Is there a way to prioritize calls in multiple queues based on hold time?  I
have three queues set up on my Asterisk PBX with agents logged into all
three queues.  I've noticed that sometimes calls in one queue will make it
through in a couple minutes while another queue will be backed up with
people having been on hold for 30+ minutes.  Is it possibly the fact that I
am set for the rrmemory ring strategy?

Thanks,

Shaun

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RE: [Asterisk-Users] Dial Command M(x) Option

2004-12-03 Thread Shaun Tierney
What version of Asterisk should I be applying this patch to?  The patch
command doesn't seem to be working.  I think because the dates on the files
in Asterisk 1.0.2 don't match the dates in the diff file.  Any ideas?  The
patch seems to work partially.  When I run patch -p4  app_dial_rev5.diff
from the asterisk-1.0.2 directory, this is what I get.

missing header for unified diff at line 8 of patch
can't find file to patch at input line 8
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|Index: pbx.c
|===
|RCS file: /usr/cvsroot/asterisk/pbx.c,v
|retrieving revision 1.173
|diff -u -r1.173 pbx.c
|--- pbx.c  19 Nov 2004 05:18:10 -  1.173
|+++ pbx.c  22 Nov 2004 22:19:48 -
--
File to patch: pbx.c
patching file pbx.c
Hunk #1 FAILED at 4935.
Hunk #2 succeeded at 4986 with fuzz 2 (offset -428 lines).
1 out of 2 hunks FAILED -- saving rejects to file pbx.c.rej
missing header for unified diff at line 174 of patch
can't find file to patch at input line 174
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|Index: include/asterisk/pbx.h
|===
|RCS file: /usr/cvsroot/asterisk/include/asterisk/pbx.h,v
|retrieving revision 1.33
|diff -u -r1.33 pbx.h
|--- include/asterisk/pbx.h 13 Nov 2004 22:44:33 -  1.33
|+++ include/asterisk/pbx.h 22 Nov 2004 22:19:48 -
--
File to patch: include/asterisk/pbx.h
patching file include/asterisk/pbx.h
Hunk #1 FAILED at 578.
1 out of 1 hunk FAILED -- saving rejects to file include/asterisk/pbx.h.rej
missing header for unified diff at line 191 of patch
can't find file to patch at input line 191
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|Index: apps/app_dial.c
|===
|RCS file: /usr/cvsroot/asterisk/apps/app_dial.c,v
|retrieving revision 1.107
|diff -u -r1.107 app_dial.c
|--- apps/app_dial.c21 Nov 2004 20:38:32 -  1.107
|+++ apps/app_dial.c22 Nov 2004 22:19:49 -
--
File to patch: apps/app_dial.c
patching file apps/app_dial.c
Hunk #1 succeeded at 67 (offset -1 lines).
Hunk #2 succeeded at 458 (offset -32 lines).
Hunk #3 FAILED at 466.
Hunk #4 succeeded at 660 (offset -8 lines).
Hunk #5 succeeded at 951 (offset -57 lines).
Hunk #6 succeeded at 1009 (offset -8 lines).
Hunk #7 succeeded at 974 (offset -57 lines).
1 out of 7 hunks FAILED -- saving rejects to file apps/app_dial.c.rej

Am I just applying the patch incorrectly or something?

Thanks for the help,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian West
Sent: Friday, December 03, 2004 10:09 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Dial Command M(x) Option


http://bugs.digium.com/bug_view_page.php?bug_id=0002905

 This email and any attached files are confidential and copyright
 protected.  If you are not the addressee, any dissemination, distribution
 or copying of this communication is strictly prohibited.  Unless otherwise
 expressly agreed in writing, nothing stated in this communication shall be
 legally binding.

FYI these types of disclaimers ARE STUPID... And crack me up when I see
them...

bkw

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[Asterisk-Users] Dial Command M(x) Option

2004-12-02 Thread Shaun Tierney
http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html

I saw this post about the M(x) option for the Dial command, but I could not
find a reply questions posed here.  I am wanting to pass the Zap channel
that the original call came from to my macro embedded in the Dial command.
I've tried to add arguments to the macro by using the syntax M(x,arg1), and
I always get the following errors.

Dec  2 15:57:53 WARNING[96234416]: app_dial.c:629 dial_exec: Could not find
macro to which we should jump.
Dec  2 15:57:53 WARNING[96234416]: app_dial.c:636 dial_exec: Macro flag set
without trailing ')

Is there a patch that would allow me to do this?  Or is there a way to get
the Zap channel name that requested the Dial on the outbound trunk from
within the macro?

Regards,

Shaun Tierney

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[Asterisk-Users] callprogress option

2004-11-22 Thread Shaun Tierney
From what I've been reading about the callprogress option, it seems like it
will work properly only with a T1 or PRI in the US.  Is that correct or are
there still issues with call progress detection even if those qualifications
are met?

Thanks,

Shaun Tierney

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RE: [Asterisk-Users] callprogress option

2004-11-22 Thread Shaun Tierney
Ok, so if I turn callprogress off, and try to connect a call which is
bridged between an incoming line and an outgoing line, will it treat the
call as being answered once it is bridged or once it is actually answered on
the outgoing T1 trunk?

Thanks,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric
Wieling
Sent: Monday, November 22, 2004 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] callprogress option


Shaun Tierney wrote:

From what I've been reading about the callprogress option, it seems like
it
 will work properly only with a T1 or PRI in the US.  Is that correct or
are
 there still issues with call progress detection even if those
qualifications
 are met?

If you ask me it doesnt' work well nomatter what kind of line you have.

VoIP (IAX/SIP/H323), PRI, and T-1/E-1 do not need callprogress.  The
telco provides everything required for progress detection.  Analog
ports don't (usually) provide this and so callprogress is needed.
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RE: [Asterisk-Users] MYSQL Dialplan Question

2004-11-17 Thread Shaun Tierney
I have already verified the permissions on the database.  I had granted all
permissions on this database to the username I am using in the dialplan.  I
used the statement GRANT ALL ON asteriskdb.* TO [EMAIL PROTECTED] IDENTIFIED
BY 'abc123';. I have logged into the MySQL console and was able to run the
UPDATE query from there using the same username and password I am trying to
use from the dialplan, so it seems to be specifically a problem with the
MYSQL addon application not being able to write or something.  Could it be
that the MYSQL application is set up for read only?  Did I miss a compile
option or something?

Thanks,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin
Brennan
Sent: Wednesday, November 17, 2004 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MYSQL Dialplan Question


If you can't update with SQL commands from the CLI then you need to check
your permissions in database mysql.
Read Mysql docs.
info mysql
MySQL Database Administration - Privilage System
Br /Kev/

- Original Message -
From: Shaun Tierney [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 10:46 PM
Subject: RE: [Asterisk-Users] MYSQL Dialplan Question


 Thanks for the help.  Downloading and installing asterisk-addons fixed my
 problem with the MYSQL application error.  Now I am having another
 difficulty though.  I am unable to update fields in the database.  I even
 hardcoded the query rather than using Asterisk dialplan variables just to
 see if that was the problem.  I am able to update fields using the MySQL
 console logging in with the same username and password I use in the
 dialplan.  Reading data seems to work great from the dialplan, just can't
 write to the database.  I'm using the following syntax.

 MYSQL(Query resultid ${connid} Update table set field=fieldvalue where
 where_expression)

 Any thoughts?

 Thanks,

 Shaun


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RE: [Asterisk-Users] MYSQL Dialplan Question

2004-11-17 Thread Shaun Tierney
Thank you very much.  That fixed my update problem.

Regards,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andreas
Sikkema
Sent: Wednesday, November 17, 2004 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MYSQL Dialplan Question


[EMAIL PROTECTED] wrote:
 Could it be that the MYSQL application is set up for read only?
 Did I miss a compile option or something?

The MYSQL Application as it is is not suited for updates
and / or inserts.

See http://lists.digium.com/pipermail/asterisk-users/2004-August/060279.html
for my patch to help this.

--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11  3011 VH Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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[Asterisk-Users] Call Status

2004-11-17 Thread Shaun Tierney
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone.  I would like to be able to detect the actual call status and
respond to it in the dialplan.  ${DIALSTATUS} just seems to tell me whether
or not the channel answered rather than the actual callee, so it equals
ANSWER every time I dial because the bridge is automatic.  Is there any
command or variable out there that will allow me to determine the actual
call status?

Thanks,

Shaun Tierney

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RE: [Asterisk-Users] MYSQL Dialplan Question

2004-11-16 Thread Shaun Tierney
Thanks for the help.  Downloading and installing asterisk-addons fixed my
problem with the MYSQL application error.  Now I am having another
difficulty though.  I am unable to update fields in the database.  I even
hardcoded the query rather than using Asterisk dialplan variables just to
see if that was the problem.  I am able to update fields using the MySQL
console logging in with the same username and password I use in the
dialplan.  Reading data seems to work great from the dialplan, just can't
write to the database.  I'm using the following syntax.

MYSQL(Query resultid ${connid} Update table set field=fieldvalue where
where_expression)

Any thoughts?

Thanks,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Carlton
O'Riley
Sent: Monday, November 15, 2004 11:03 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] MYSQL Dialplan Question


You need to download the asterisk-addons to have mysql support now.  It was
only moved to its own project due to licensing changes with MySQL.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Shaun Tierney
 Sent: Monday, November 15, 2004 11:56 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] MYSQL Dialplan Question

 Well, I looked at the makefile and I could not see any
 options for SQL.  So I did a grep for SQL on the distribution
 files and found that in version 0.7.0 support for MySQL was
 removed, so I'm guessing I'm just going to have to switch to
 Postgres or something.

 Thanks for the help,

 Shaun

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 Steven Critchfield
 Sent: Monday, November 15, 2004 10:11 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] MYSQL Dialplan Question


 On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote:
  I am new to Asterisk, and I am having trouble connecting to
 the MySQL
  database located on the same machine as my Asterisk box.  When the
 dialplan
  tried to connect to MySQL database, I get the following
 error message
  on
 the
  Asterisk console.
 
  Nov 15 09:29:41 WARNING[39760]: pbx.c:1279
 pbx_extension_helper:
  No application 'MYSQL' for extension (default, s, 5)
 
  Here is the corresponding line in my dialplan.
 
  exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb)
 
  I am able to connect to the database with the same username and
  password using the MySQL console.  Am I missing something in my
  installation or configuration that would cause this?

 Yes you are missing something in your installation. From
 reading the error message I can see you did not build the
 mysql application. My guess would be that it is not enabled
 in the makefile. Go edit it, compile, install, and try again.
 --
 Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] MYSQL Dialplan Question

2004-11-15 Thread Shaun Tierney
I am new to Asterisk, and I am having trouble connecting to the MySQL
database located on the same machine as my Asterisk box.  When the dialplan
tried to connect to MySQL database, I get the following error message on the
Asterisk console.

Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No
application 'MYSQL' for extension (default, s, 5)

Here is the corresponding line in my dialplan.

exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb)

I am able to connect to the database with the same username and password
using the MySQL console.  Am I missing something in my installation or
configuration that would cause this?

Thanks,

Shaun

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RE: [Asterisk-Users] MYSQL Dialplan Question

2004-11-15 Thread Shaun Tierney
Well, I looked at the makefile and I could not see any options for SQL.  So
I did a grep for SQL on the distribution files and found that in version
0.7.0 support for MySQL was removed, so I'm guessing I'm just going to have
to switch to Postgres or something.

Thanks for the help,

Shaun

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Monday, November 15, 2004 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MYSQL Dialplan Question


On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote:
 I am new to Asterisk, and I am having trouble connecting to the MySQL
 database located on the same machine as my Asterisk box.  When the
dialplan
 tried to connect to MySQL database, I get the following error message on
the
 Asterisk console.

 Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No
 application 'MYSQL' for extension (default, s, 5)

 Here is the corresponding line in my dialplan.

 exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb)

 I am able to connect to the database with the same username and password
 using the MySQL console.  Am I missing something in my installation or
 configuration that would cause this?

Yes you are missing something in your installation. From reading the
error message I can see you did not build the mysql application. My
guess would be that it is not enabled in the makefile. Go edit it,
compile, install, and try again.
--
Steven Critchfield [EMAIL PROTECTED]

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