Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

[Simith Nambiar]

Doug Lytle wrote:
 Simith Nambiar wrote:
   
 Hello Doug,
  Thank you for your response, if you see my 
 e-mail above, i wanted the Read to happen after the Call is connected 
 (i.e after Dial
 


 Pen and paper?

 Seriously,  I thought you were just having trouble elucidating your thoughts. 
  

 Once a channel has been bridged, you either can transfer that caller to an 
 IVR to collect the digits or have the recipient callee collect the 
 information.
   
[Simith] I could work out a solution using the application map feature 
in features.conf  to collect the digits from the Callee, Maybe there is 
a better way, but this works for me !

calleedigitcollect1 = 1,self/callee,Macro,redirect-caller1


Thanks to the people who built Asterisk, looks like anything is possible :-)
   
 Doug


   

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Re: [asterisk-users] CDR Desgin

[Grey Man]

On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy [EMAIL PROTECTED] wrote:
 For the moment, let's not worry about the implementation. Let's
 get consensus on the spec first. In the scenario, where A calls B,
 B xfers A to C, C xfers A to D, or some such similar scenario,
 half the world wants a single CDR for A, from the time it started,
 to the time it hung up with D. The other half wants A-B's dial and
 bridge,
 a cdr for A  C's bridge, a CDR for A  D's bridge, and mayhaps some
 CDRs
 to describe the xfers, where B xfers A to C and C xfers A to D.

 My document is pointing in the former direction, and either we need to
 spec both, or pick one.

To me the obvious answer is to provide a CDR for every call leg so for
A calling B via Asterisk there would be two CDRs produced. It's far
far easier to disregard the unwanted CDRs than it is to try and
generate the missing ones and in some cases it's virtually impossible.
If it's weighed up I think people would vote to have accurate CDRs
ahead of anything else and if single legs are the best way to do that
then it's the way it should be done.

In addition with single leg CDRs it will solve the dilemna about
acommodating every possible call scenario that I know has caused you a
lot of consternation over the last 18 months.

Sure it's a change from the current situation so maybe needs to use
the standard apporach of a configuration setting to opt in initially
before becoming the default in the subsequent major release.

Regards,

Greyman.

P.S. Sorry about the duplicate post. I actually sent the email to the
list 4 times with around 8 hour spacings and I'm not sure why there
was a problem getting it through.

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[asterisk-users] OSLEC build errors on DAHDI [was: Re: HPEC performance]

[Tzafrir Cohen]

On Wed, Nov 19, 2008 at 03:46:35PM -0700, Joseph L. Casale wrote:
 Not trivial but not as voodoo as before:
 
   http://docs.tzafrir.org.il/dahdi-linux/#_oslec
 
 Tzafrir,
 I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now
 when compiling I get the following:
 WARNING: oslec_create 
 [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined!
 WARNING: oslec_free [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] 
 undefined!
 WARNING: oslec_update 
 [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined!
 
 Any ideas?

Try:

  echo 'obj-m += echo.o' drivers/staging/echo/Kbuild

And re-run 'make'

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

[Atis Lezdins]

On Tue, Nov 25, 2008 at 2:19 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
 On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
 On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
 wrote:
  I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
  and tools but my calls aren't logged. I'd enabled mysql log and
  noticed that asterisk send a 'DESC cdr' so connection is working
  between asterisk and mysql and I am able to call other phones so
  Asterisk is working as well. No error messages on startup though.
 
  Any idea why is it happen? As I realized there is some differences
  between 1.2 (my previous system) and 1.6 log system.

 I suspect that you have some unique index on the table which is
 conflicting with the inserted fields.  Once you figure out which field is
 causing the conflict, it should be easier to figure out where the problem
 actually lies.

 You should also check Asterisk log for warnings. 1.6 should detect
 table structure and warn about missing fields. If it's so, perhaps you
 can change asterisk - mysql (res_cdr_addon_mysql if i remember
 correctly) to do an alter on your table - then it will automagically
 create missing fields.

 You remember incorrectly.  None of the CDR drivers currently have the
 capability to alter tables.  What they will do is to adapt to the table
 structure and insert only the required fields.  Only realtime table drivers
 have the capability of altering tables and then, only if you turn that
 behavior on.  By default, Asterisk does not alter table structures.


Oh, my mistake :)

 BTW, if you 'core set debug 2 cdr_addon_mysql.c' (and make sure debug is
 enabled to the console, via /etc/asterisk/logger.conf), then the SQL will be
 printed to the console.  That should help you find where the problem lies.

Actually SQL's are logged with debug 1, debug 2 is too much of everything.

I didn't knew that you could use filename.c when setting debug
level, seems quite useful :)

Regards,
Atis


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] Set a specific registration expiry value to a given peer without touching defaultexpiry in sip.conf ?

[Olivier]

Hi,

I've got several trunks in my 1.6.0.1 setup.
One of them is asking for 1800 sec registrations.

You can provide this value setting defaultexpiry to 1800 in sip.conf but how
can you specify this duration to this specific trunk and not affect the
others ?
An option to register statement in sip.conf would be perfect ...

Regards
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[asterisk-users] Disabling Call-Waiting

[Elliot Murdock]

Hello!

I have a few sip devices and it is necessary for me to disable call-waiting
and immediately return a busy signal if the sip's channel is busy on them.

What is the procedure to do so in Asterisk 1.4?

Thank you,

Elliot
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Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

[Artifex Maximus]

Hello!

On Tue, Nov 25, 2008 at 2:25 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
 On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote:
 On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
  On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]

 wrote:
   I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
   and tools but my calls aren't logged. I'd enabled mysql log and
   noticed that asterisk send a 'DESC cdr' so connection is working
   between asterisk and mysql and I am able to call other phones so
   Asterisk is working as well. No error messages on startup though.
  
   Any idea why is it happen? As I realized there is some differences
   between 1.2 (my previous system) and 1.6 log system.

 I suspect that you have some unique index on the table which is
 conflicting with the inserted fields.  Once you figure out which field is
 causing the conflict, it should be easier to figure out where the problem
 actually lies.
All indexes are simple indexes so that cannot be a problem.
Asterisk's log did not noticed any problem with table structure.

[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got hostname of localhost
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got port of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a timeout of 0
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got sock file of
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got user of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got dbname of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got password of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Not running in
calldate compatibility mode
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Successfully
connected to MySQL database.
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'calldate' of type 'datetime'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
calldate, type datetime
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'clid'
of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
clid, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'src' of
type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
src, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'dst' of
type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dst, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'dcontext' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dcontext, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'channel' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
channel, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'dstchannel' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dstchannel, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'lastapp' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
lastapp, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'lastdata' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
lastdata, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'duration' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
duration, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'billsec' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
billsec, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'disposition' of type 'varchar(45)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
disposition, type varchar(45)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'amaflags' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
amaflags, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'accountcode' of type 'varchar(20)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
accountcode, type varchar(20)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'uniqueid' of type 'varchar(32)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
uniqueid, type varchar(32)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'userfield' of type 'varchar(255)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
userfield, type varchar(255)
[Nov 21 16:05:16] VERBOSE[10740] logger.c: [Nov 21 16:05:16]
cdr_addon_mysql.so = (MySQL CDR Backend)

 BTW, if you 'core set debug 2 cdr_addon_mysql.c' (and make sure debug is
 enabled to the console, via 

Re: [asterisk-users] Disabling Call-Waiting

[Sebastian]

You can try call-limit = 1 in sip.conf for each phone.

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Elliot Murdock
Sent: martes, 25 de noviembre de 2008 11:04 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Disabling Call-Waiting

 

Hello!

I have a few sip devices and it is necessary for me to disable call-waiting
and immediately return a busy signal if the sip's channel is busy on them. 

What is the procedure to do so in Asterisk 1.4?

Thank you,

Elliot

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Re: [asterisk-users] Disabling Call-Waiting

[Elliot Murdock]

Hello Sebastian,
Thanks!
However, I tried using call-limit and the phones are still getting the
calls.  Maybe I am doing something wrong.
Elliot
On Tue, Nov 25, 2008 at 3:14 PM, Sebastian [EMAIL PROTECTED] wrote:

  You can try call-limit = 1 in sip.conf for each phone.





 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Elliot Murdock
 *Sent:* martes, 25 de noviembre de 2008 11:04 a.m.
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Disabling Call-Waiting



 Hello!

 I have a few sip devices and it is necessary for me to disable call-waiting
 and immediately return a busy signal if the sip's channel is busy on them.

 What is the procedure to do so in Asterisk 1.4?

 Thank you,

 Elliot

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Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

[Sebastian]

I think call-limit is for 1 outbound and 1 inbound so if you get a call you
wont get other but if you do an outbound call an incoming call will be
allowed.
Maybe you can configure 1 line in your phone or try Check device state
before make the call in extensions.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: martes, 25 de noviembre de 2008 11:25 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

Hello!

On Tue, Nov 25, 2008 at 2:25 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
 On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote:
 On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
  On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]

 wrote:
   I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
   and tools but my calls aren't logged. I'd enabled mysql log and
   noticed that asterisk send a 'DESC cdr' so connection is working
   between asterisk and mysql and I am able to call other phones so
   Asterisk is working as well. No error messages on startup though.
  
   Any idea why is it happen? As I realized there is some differences
   between 1.2 (my previous system) and 1.6 log system.

 I suspect that you have some unique index on the table which is
 conflicting with the inserted fields.  Once you figure out which field is
 causing the conflict, it should be easier to figure out where the problem
 actually lies.
All indexes are simple indexes so that cannot be a problem.
Asterisk's log did not noticed any problem with table structure.

[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got hostname of localhost
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got port of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a timeout of 0
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got sock file of
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got user of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got dbname of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got password of xxx
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Not running in
calldate compatibility mode
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Successfully
connected to MySQL database.
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'calldate' of type 'datetime'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
calldate, type datetime
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'clid'
of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
clid, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'src' of
type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
src, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field 'dst' of
type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dst, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'dcontext' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dcontext, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'channel' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
channel, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'dstchannel' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
dstchannel, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'lastapp' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
lastapp, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'lastdata' of type 'varchar(80)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
lastdata, type varchar(80)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'duration' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
duration, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'billsec' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
billsec, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'disposition' of type 'varchar(45)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
disposition, type varchar(45)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'amaflags' of type 'int(11)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
amaflags, type int(11)
[Nov 21 16:05:16] DEBUG[10740] cdr_addon_mysql.c: Got a field
'accountcode' of type 'varchar(20)'
[Nov 21 16:05:16] NOTICE[10740] cdr_addon_mysql.c: Found a DB column
accountcode, type varchar(20)
[Nov 21 16:05:16] DEBUG[10740] 

Re: [asterisk-users] Disabling Call-Waiting

[Gordon Henderson]

On Tue, 25 Nov 2008, Elliot Murdock wrote:

 Hello!

 I have a few sip devices and it is necessary for me to disable call-waiting
 and immediately return a busy signal if the sip's channel is busy on them.

 What is the procedure to do so in Asterisk 1.4?

Read the phone manual and work out how to disable it in the phone.

That's the easiest way I've found, anyway...

Gordon

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[asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6

[Jason Lixfeld]

This link 
(http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/
 
) seems to indicate that in order to upgrade AsteriskNOW v1.5 from  
Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package.

Does anyone know where to find that upgrade package?  If it doesn't  
yet exist, what is the process for upgrading?

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Re: [asterisk-users] The sound is played but I did not hear

[jhon digital21]

there is no answer ???

2008/11/12 jhon digital21 [EMAIL PROTECTED]

 Hello,

 I have another little problem with my ZAPs channels, in fact, when I
 received a call, I heard no sound while in the CLI, sound is played:

 -- Starting simple switch on 'Zap/4-1'
 -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack
 -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, hello-world) in new
 stack
 -- Zap/4-1 Playing 'hello-world' (language 'en')
 **
 -- Executing [EMAIL PROTECTED]:8] WaitExten(Zap/4-1, ) in new stack
 == Spawn extension (from-zaptel, s, exited non-zero on 'Zap/4-1'

 -- Hungup 'Zap/4-1'

 Here is my setup:

 Zapata.conf:

 ;
 ; Zapata telephony interface
 ;
 ; Configuration file

 [trunkgroups]

 [channels]

 language=en
 context=from-zaptel
 signalling=fxs_ks
 rxwink=300 ; Atlas seems to use long (250ms) winks
 ;
 ; Whether or not to do distinctive ring detection on FXO lines
 ;
 ;usedistinctiveringdetection=yes

 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=no
 ;echotraining=800
 rxgain=0.0
 txgain=0.0
 group=0
 callgroup=1
 pickupgroup=1
 immediate=no

 busydetect=yes
 busycount=5



 ;faxdetect=both
 faxdetect=incoming
 ;faxdetect=outgoing
 ;faxdetect=no

 ;Include genzaptelconf configs
 #include zapata-auto.conf

 group=1

 ;Include AMP configs

 #include zapata_additional.conf

 Zaptel.conf:

 # Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 (MASTER)
 fxoks=1
 fxsks=4

 # Global data

 loadzone = us
 defaultzone = us


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Re: [asterisk-users] The sound is played but I did not hear

[Doug Lytle]

jhon digital21 wrote:
 there is no answer ???



Try putting a Wait(1) before playing the sound effect.

Doug


-- 
 
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Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Disabling Call-Waiting

[Mikhail (Plus Plus)]

Sorry for hijacking this thread, but I need something similar in 
opposite way the original poster wants:

I have real incoming phone number and all calls on this phone number are 
redirected to local extension: 1234567 - 515
where 1234567 is a phone number and 515 is a local extension in 
asterisk. The phone number I get from my provider supposedly has 2 
incoming lines. What I want is:
If 515 extension is busy, redirect call to local extension 516.
How this can be achieved?

Right now I have:

exten = 1234567,1,Dial(SIP/515)
exten = s-NOANSWER,n,Dial(SIP/516)
exten = s-BUSY,n,Dial(SIP/516)

But this does not work (I guess because I'm wrong).

Thanks.

Gordon Henderson wrote:
 On Tue, 25 Nov 2008, Elliot Murdock wrote:
 
 Hello!

 I have a few sip devices and it is necessary for me to disable call-waiting
 and immediately return a busy signal if the sip's channel is busy on them.

 What is the procedure to do so in Asterisk 1.4?
 
 Read the phone manual and work out how to disable it in the phone.
 
 That's the easiest way I've found, anyway...
 
 Gordon
 
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Re: [asterisk-users] AsteriskNOW 1.5 upgrade from 1.4 to 1.6

[Jason Parker]

Jason Lixfeld wrote:
 This link 
 (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/
  
 ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from  
 Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package.
 
 Does anyone know where to find that upgrade package?  If it doesn't  
 yet exist, what is the process for upgrading?
 

Haven't figured out quite how I want to do this yet, but this is what has worked
for me in testing (you may need to modify this slightly to add asterisk addons,
if you're using it).


Run `yum shell`, then in that shell, execute:

install asterisk16-core asterisk16
remove asterisk14 asterisk14-core
ts solve
ts run
remove asterisk14-core
ts solve
ts run


If it went properly, it won't try to remove anything like FreePBX (it will
prompt you before it does anything, so you can say 'No' if it tries).

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Re: [asterisk-users] Disabling Call-Waiting

[Gordon Henderson]

On Tue, 25 Nov 2008, Mikhail (Plus Plus) wrote:

 Sorry for hijacking this thread, but I need something similar in
 opposite way the original poster wants:

 I have real incoming phone number and all calls on this phone number are
 redirected to local extension: 1234567 - 515
 where 1234567 is a phone number and 515 is a local extension in
 asterisk. The phone number I get from my provider supposedly has 2
 incoming lines. What I want is:
 If 515 extension is busy, redirect call to local extension 516.
 How this can be achieved?

 Right now I have:

 exten = 1234567,1,Dial(SIP/515)
 exten = s-NOANSWER,n,Dial(SIP/516)
 exten = s-BUSY,n,Dial(SIP/516)

 But this does not work (I guess because I'm wrong).

If you have programmed the phones to only accept 1 incoming call (ie. 
turned of call-waiting on the phones thmselves), then simply:

   exten = 1234567,1,Dial(SIP/515)
   exten = 1234567,n,Dial(SIP/516)

Will work.

You may want to put a time-out on the first Dial to drop into the 2nd one 
if the first phone doesn't answer, otherwise it'll ring until the caller 
hangs up.

Gordon



 Thanks.

 Gordon Henderson wrote:
 On Tue, 25 Nov 2008, Elliot Murdock wrote:

 Hello!

 I have a few sip devices and it is necessary for me to disable call-waiting
 and immediately return a busy signal if the sip's channel is busy on them.

 What is the procedure to do so in Asterisk 1.4?

 Read the phone manual and work out how to disable it in the phone.

 That's the easiest way I've found, anyway...

 Gordon

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Re: [asterisk-users] Disabling Call-Waiting

[Sebastian]

You can try:

exten = 1234567,1,Dial(SIP/515)
exten = 1234567,101,Dial(SIP/516)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mikhail (Plus
Plus)
Sent: martes, 25 de noviembre de 2008 02:08 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disabling Call-Waiting

Sorry for hijacking this thread, but I need something similar in 
opposite way the original poster wants:

I have real incoming phone number and all calls on this phone number are 
redirected to local extension: 1234567 - 515
where 1234567 is a phone number and 515 is a local extension in 
asterisk. The phone number I get from my provider supposedly has 2 
incoming lines. What I want is:
If 515 extension is busy, redirect call to local extension 516.
How this can be achieved?

Right now I have:

exten = 1234567,1,Dial(SIP/515)
exten = s-NOANSWER,n,Dial(SIP/516)
exten = s-BUSY,n,Dial(SIP/516)

But this does not work (I guess because I'm wrong).

Thanks.

Gordon Henderson wrote:
 On Tue, 25 Nov 2008, Elliot Murdock wrote:
 
 Hello!

 I have a few sip devices and it is necessary for me to disable
call-waiting
 and immediately return a busy signal if the sip's channel is busy on
them.

 What is the procedure to do so in Asterisk 1.4?
 
 Read the phone manual and work out how to disable it in the phone.
 
 That's the easiest way I've found, anyway...
 
 Gordon
 
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__ Information from ESET Smart Security, version of virus signature
database 2911 (20080229) __

The message was checked by ESET Smart Security.

http://www.eset.com

 

__ Information from ESET Smart Security, version of virus signature
database 2911 (20080229) __

The message was checked by ESET Smart Security.

http://www.eset.com
 


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Re: [asterisk-users] Disabling Call-Waiting

[Elliot Murdock]

Hello!

Thanks for the responses.  I'll look into the phone devices themselves.

I am wondering, if call-limit did not solve my problem, what is the
call-limit parameter supposed to do anyway?

Later,

Elliot

On Tue, Nov 25, 2008 at 6:29 PM, Sebastian [EMAIL PROTECTED] wrote:

 You can try:

 exten = 1234567,1,Dial(SIP/515)
 exten = 1234567,101,Dial(SIP/516)

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mikhail
 (Plus
 Plus)
 Sent: martes, 25 de noviembre de 2008 02:08 p.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Disabling Call-Waiting

 Sorry for hijacking this thread, but I need something similar in
 opposite way the original poster wants:

 I have real incoming phone number and all calls on this phone number are
 redirected to local extension: 1234567 - 515
 where 1234567 is a phone number and 515 is a local extension in
 asterisk. The phone number I get from my provider supposedly has 2
 incoming lines. What I want is:
 If 515 extension is busy, redirect call to local extension 516.
 How this can be achieved?

 Right now I have:

 exten = 1234567,1,Dial(SIP/515)
 exten = s-NOANSWER,n,Dial(SIP/516)
 exten = s-BUSY,n,Dial(SIP/516)

 But this does not work (I guess because I'm wrong).

 Thanks.

 Gordon Henderson wrote:
  On Tue, 25 Nov 2008, Elliot Murdock wrote:
 
  Hello!
 
  I have a few sip devices and it is necessary for me to disable
 call-waiting
  and immediately return a busy signal if the sip's channel is busy on
 them.
 
  What is the procedure to do so in Asterisk 1.4?
 
  Read the phone manual and work out how to disable it in the phone.
 
  That's the easiest way I've found, anyway...
 
  Gordon
 
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 __ Information from ESET Smart Security, version of virus signature
 database 2911 (20080229) __

 The message was checked by ESET Smart Security.

 http://www.eset.com



 __ Information from ESET Smart Security, version of virus signature
 database 2911 (20080229) __

 The message was checked by ESET Smart Security.

 http://www.eset.com



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Re: [asterisk-users] Disabling Call-Waiting

[Tilghman Lesher]

On Tuesday 25 November 2008 10:07:34 Mikhail (Plus Plus) wrote:
 Sorry for hijacking this thread, but I need something similar in
 opposite way the original poster wants:

 I have real incoming phone number and all calls on this phone number are
 redirected to local extension: 1234567 - 515
 where 1234567 is a phone number and 515 is a local extension in
 asterisk. The phone number I get from my provider supposedly has 2
 incoming lines. What I want is:
 If 515 extension is busy, redirect call to local extension 516.
 How this can be achieved?

 Right now I have:

 exten = 1234567,1,Dial(SIP/515)
 exten = s-NOANSWER,n,Dial(SIP/516)
 exten = s-BUSY,n,Dial(SIP/516)

 But this does not work (I guess because I'm wrong).

You're missing a step:

exten = 1234567,1,Dial(SIP/515)
exten = 1234567,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(SIP/516)
exten = s-BUSY,1,Dial(SIP/516)

or better:

exten = 1234567,1,Dial(SIP/515,30)
exten = 1234567,n,GotoIf($[${DIALSTATUS}=NOANSWER]?failover)
exten = 1234567,n,GotoIf($[${DIALSTATUS}=BUSY]?failover)
exten = 1234567,n,Goto(voicemail)
exten = 1234567,n(failover),Dial(SIP/516,30)
exten = 1234567,n(voicemail),Voicemail(515,u)

-- 
Tilghman

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Re: [asterisk-users] CDR Design

[Anthony Francis]

We are suggesting the same thing, what you describe is multidimensional. 
If you think of the cdr's as being in a database, and say you wanted to 
have it show you all the calls today and all the calls that are 
associated with that call. Your select grabs the first dimension, a list 
of all calls. Then using the unique identifier of each call you build a 
second dimension of the related calls.

[EMAIL PROTECTED] wrote:
 In order to avoid a multidimensional schema we could have 1 cdr per call 
 leg. So , for instance, one
 call that had 3 different dial() commands as outgoing attempts would be 
 described by 4
 CDRs (1 for the incoming leg that has all the originating channel data 
 and 3 for the outgoing
 legs that hold all the terminating channel's data). Those CDRs would be 
 bound by a unique
 identifier field (the same for all 4). The terminating CDRs could be 
 also identified by a increment field that indicates
 the order that the channels were called. Another issue is that failed 
 attempts should also be logged because
 this is valuable info for many (or at least have the option to choose 
 the desired behavior - which is available in asterisk as we speak).

 Anthony Francis wrote:
   
 It is my belief that before redesigning the CDR engine some time should 
 be spent looking at current PSTN CDR formats and what information is 
 kept in them.
 The main problem that I feel we face is that calls can be complicated, 
 but we want the record of it to not be.
 In reality a CDR that incorporates all information about a call would 
 have at least two dimensions.
 In the first you would have the base call record as we do now, in the 
 second we would have the event list.
 The event list would be a time indexed list of event names and 
 attributes, just as you would currently store event information.
 The event list would be your glue (with a bit of front end logic of 
 course.) that would relate a call that dialed X external numbers to the 
 X different new CDR's that generated.
 That would allow you all the call path info you could ever want. The 
 most important thing would be a new config file that allows an 
 administrator granular control over what information is important to 
 them. And of course a keep it simple stupid mode that just writes the 
 top level cdr as it does now.

 [EMAIL PROTECTED] wrote:
   
 
 I think that the custom cdr back-end can be successfully used to 
 maximize or minimize the CDRs detailing
 on a per-needs basis. Furthermore, the CDR() function gives plenty of 
 room for even more detailing.
 In my opinion the detail level (fields) is not the issue with the CDRs 
 generation nor is the lack of backends (asterisk gives a lot of different
 backends to store your CDRs). I find the issue with asterisk CDRs to be 
 in the lack of proper CDRs generation for the B-leg of every call.
 If we want to really track what happens during a call through the CDRs 
 one has to have all the details not only for the incoming channel
 but for the outgoing one as well. Furthermore, one needs to be able to 
 tweak the B-leg CDRs like he does with the incoming legs. So what
 needs to be done in my opinion is record every B-leg CDR when such an 
 event occurs and give the user access to the CDR info by
 extending the CDR() function (so that one can specify the channel of the 
 CDR that is being tweaked) or creating a seperate one for
 the outgoing channels.

 Grey Man wrote:
   
 
   
 I've taken the liberty of starting a new thread to discuss the design
 of the Asterisk CDR mechanism. The discussion has been kindly
 initiated by murf putting together a proposal (link ommitted to see if
 email gets accepted).

 After reading the proposal I still don't think it's the right way to
 go. To my mind adding more channel variables increases the complexity
 in a situation that is already overly so. I think it's a mistake to
 try and think about all the different call scenarios and come up with
 little tricks for the more complicated ones. There will always be
 something missed; app_shotgun initiates calls to 100 random numbers
 and as soon as three or more calls are answered it will start randonly
 transferring them amongst each other at 2 second intervals.

 I think it's important to clarify at the outset what a CDR should be.
 The most fundamental requirement for CDRs is that they accurately
 record the following pieces of information for EVERY call entering or
 leaving the system (note every means every and not; channel calls
 but not peer calls).

 1. Destination (aka as A Number)
 2. AccountCode (aka as B Number)
 3. Call Start Time (answer time),
 4. Duration.

 Of course adding extra information can be very useful and I'm not
 proposing any fields be removed from the current implementation
 (although for pity's sake one change that should be made it to use a
 GUID/UUID for the CDR's uniqueid and save endless confusion).

 People that really do need verbose or enhanced CDRs to do things 

Re: [asterisk-users] Disabling Call-Waiting

[Tilghman Lesher]

On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
 Thanks for the responses.  I'll look into the phone devices themselves.

 I am wondering, if call-limit did not solve my problem, what is the
 call-limit parameter supposed to do anyway?

The call-limit is actually kind of deprecated.  Using the GROUP() function
is now the preferred way to do this.

exten = 1234,1,Set(GROUP()=foo)
exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
exten = 1234,n,DIal(SIP/tilghman,30)
exten = 1234,n(voicemail),Voicemail(1234,
${IF($[${DIALSTATUS}=BUSY]?b:u)})

-- 
Tilghman

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Re: [asterisk-users] CDR Desgin

[Steve Murphy]

On Tue, 2008-11-25 at 08:06 +, Grey Man wrote:
 On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy [EMAIL PROTECTED] wrote:
  For the moment, let's not worry about the implementation. Let's
  get consensus on the spec first. In the scenario, where A calls B,
  B xfers A to C, C xfers A to D, or some such similar scenario,
  half the world wants a single CDR for A, from the time it started,
  to the time it hung up with D. The other half wants A-B's dial and
  bridge,
  a cdr for A  C's bridge, a CDR for A  D's bridge, and mayhaps some
  CDRs
  to describe the xfers, where B xfers A to C and C xfers A to D.
 
  My document is pointing in the former direction, and either we need to
  spec both, or pick one.
 
 To me the obvious answer is to provide a CDR for every call leg so for
 A calling B via Asterisk there would be two CDRs produced. It's far
 far easier to disregard the unwanted CDRs than it is to try and
 generate the missing ones and in some cases it's virtually impossible.
 If it's weighed up I think people would vote to have accurate CDRs
 ahead of anything else and if single legs are the best way to do that
 then it's the way it should be done.
 
 In addition with single leg CDRs it will solve the dilemna about
 acommodating every possible call scenario that I know has caused you a
 lot of consternation over the last 18 months.
 
 Sure it's a change from the current situation so maybe needs to use
 the standard apporach of a configuration setting to opt in initially
 before becoming the default in the subsequent major release.


OK, Greyman, your logic is solid. If we provide a CDR implementation
that just generates the individual call legs, and ties them together via
the linkedid
(see team/group/newcdr), then both camps should be able to derive the
info
they need for billing, via hopefully not-overly-complex SQL queries to a
backend db.

I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of
shift.
And, yes, the implementation will make this new approach optional, and
not
default. But, pardon if I make it available via the CEL domain come
implementation time.


It should take me a week to rehash my document; perhaps longer (I'm in
bugfix mode, and this borderline development work); in the meantime,
those with decided CDR needs might make their wishes known, if they do
not think this approach will work. Speak now, or forever hold your
peace; or forever complain... or whatever.
If this is particularly distressing to you, perhaps your fears might be
slightly assuaged when you read the details...

murf


-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] CDR Design

[Steve Murphy]

On Mon, 2008-11-24 at 09:12 -0700, Anthony Francis wrote:
 It is my belief that before redesigning the CDR engine some time should 
 be spent looking at current PSTN CDR formats and what information is 
 kept in them.
 The main problem that I feel we face is that calls can be complicated, 
 but we want the record of it to not be.
 In reality a CDR that incorporates all information about a call would 
 have at least two dimensions.
 In the first you would have the base call record as we do now, in the 
 second we would have the event list.
 The event list would be a time indexed list of event names and 
 attributes, just as you would currently store event information.
 The event list would be your glue (with a bit of front end logic of 
 course.) that would relate a call that dialed X external numbers to the 
 X different new CDR's that generated.
 That would allow you all the call path info you could ever want. The 
 most important thing would be a new config file that allows an 
 administrator granular control over what information is important to 
 them. And of course a keep it simple stupid mode that just writes the 
 top level cdr as it does now.
 

Well, the CEL stuff definitely generates the event lists, and can do it
to any of the normal database backends. But Brian Degenhardt pointed out
that really, such event-list databases are practically useless. Queries
for even simple things would involve incredibly complicated queries in
terms of unique ID, event order, etc. etc.

So, a CDR backend that lists all the call legs and the info 'behind'
each leg, like what prompted this leg (an xfer vs a dial, etc), should
be sufficient to answer most business-logic questions without ambiguity,
and give you the info you seek, like how long a leg lasted. (which might
be pretty interesting to calculate, if all you have are individual
Start, answer, end, xfer, park, conf, etc. events in your db.

Brian took all the raw events to a backend, but then post-processed them
to form the records he needed.

I plan the same for CEL, but am searching for consensus on what to
generate.

murf


-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] MOH Realtime

[Sebastian]

Anybody was able to set it up??

I can't make it work, any idea??

 

Ast 1.6.0.1

 

 

Thanks

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Re: [asterisk-users] Disabling Call-Waiting

[Sebastian]

If is deprecated how do you treat a queue (realtime), that has to have just
one call for agent??

Thanks



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: martes, 25 de noviembre de 2008 03:37 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disabling Call-Waiting

On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
 Thanks for the responses.  I'll look into the phone devices themselves.

 I am wondering, if call-limit did not solve my problem, what is the
 call-limit parameter supposed to do anyway?

The call-limit is actually kind of deprecated.  Using the GROUP() function
is now the preferred way to do this.

exten = 1234,1,Set(GROUP()=foo)
exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
exten = 1234,n,DIal(SIP/tilghman,30)
exten = 1234,n(voicemail),Voicemail(1234,
${IF($[${DIALSTATUS}=BUSY]?b:u)})

-- 
Tilghman

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__ Information from ESET Smart Security, version of virus signature
database 2911 (20080229) __

The message was checked by ESET Smart Security.

http://www.eset.com

 

__ Information from ESET Smart Security, version of virus signature
database 2911 (20080229) __

The message was checked by ESET Smart Security.

http://www.eset.com
 


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Re: [asterisk-users] Disabling Call-Waiting

[Yehavi Bourvine]

Try looking at the DEV_STATE function (available separately on
Asterisk-1.4). It will tell you the status of the phone before you call the
Dial() application.

  __Yehavi:

2008/11/25 Sebastian [EMAIL PROTECTED]

 If is deprecated how do you treat a queue (realtime), that has to have just
 one call for agent??

 Thanks



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
 Lesher
 Sent: martes, 25 de noviembre de 2008 03:37 p.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Disabling Call-Waiting

  On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
  Thanks for the responses.  I'll look into the phone devices themselves.
 
  I am wondering, if call-limit did not solve my problem, what is the
  call-limit parameter supposed to do anyway?

 The call-limit is actually kind of deprecated.  Using the GROUP() function
 is now the preferred way to do this.

 exten = 1234,1,Set(GROUP()=foo)
 exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
 exten = 1234,n,DIal(SIP/tilghman,30)
 exten = 1234,n(voicemail),Voicemail(1234,
 ${IF($[${DIALSTATUS}=BUSY]?b:u)})

 --
 Tilghman

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Re: [asterisk-users] CDR Design

[[EMAIL PROTECTED]]

Yes, I know we are suggesting the same thing... I just thought you are 
suggesting putting this multidimensional
CDR in one row (which of course requires data structure other than a 
simple comma separated row - XML perhaps).
I did not understand you were referring to a conceptual 
multi-dimensional and not an actual multidimensional storage method.

Anthony Francis wrote:
 We are suggesting the same thing, what you describe is multidimensional. 
 If you think of the cdr's as being in a database, and say you wanted to 
 have it show you all the calls today and all the calls that are 
 associated with that call. Your select grabs the first dimension, a list 
 of all calls. Then using the unique identifier of each call you build a 
 second dimension of the related calls.

 [EMAIL PROTECTED] wrote:
   
 In order to avoid a multidimensional schema we could have 1 cdr per call 
 leg. So , for instance, one
 call that had 3 different dial() commands as outgoing attempts would be 
 described by 4
 CDRs (1 for the incoming leg that has all the originating channel data 
 and 3 for the outgoing
 legs that hold all the terminating channel's data). Those CDRs would be 
 bound by a unique
 identifier field (the same for all 4). The terminating CDRs could be 
 also identified by a increment field that indicates
 the order that the channels were called. Another issue is that failed 
 attempts should also be logged because
 this is valuable info for many (or at least have the option to choose 
 the desired behavior - which is available in asterisk as we speak).

 Anthony Francis wrote:
   
 
 It is my belief that before redesigning the CDR engine some time should 
 be spent looking at current PSTN CDR formats and what information is 
 kept in them.
 The main problem that I feel we face is that calls can be complicated, 
 but we want the record of it to not be.
 In reality a CDR that incorporates all information about a call would 
 have at least two dimensions.
 In the first you would have the base call record as we do now, in the 
 second we would have the event list.
 The event list would be a time indexed list of event names and 
 attributes, just as you would currently store event information.
 The event list would be your glue (with a bit of front end logic of 
 course.) that would relate a call that dialed X external numbers to the 
 X different new CDR's that generated.
 That would allow you all the call path info you could ever want. The 
 most important thing would be a new config file that allows an 
 administrator granular control over what information is important to 
 them. And of course a keep it simple stupid mode that just writes the 
 top level cdr as it does now.

 [EMAIL PROTECTED] wrote:
   
 
   
 I think that the custom cdr back-end can be successfully used to 
 maximize or minimize the CDRs detailing
 on a per-needs basis. Furthermore, the CDR() function gives plenty of 
 room for even more detailing.
 In my opinion the detail level (fields) is not the issue with the CDRs 
 generation nor is the lack of backends (asterisk gives a lot of different
 backends to store your CDRs). I find the issue with asterisk CDRs to be 
 in the lack of proper CDRs generation for the B-leg of every call.
 If we want to really track what happens during a call through the CDRs 
 one has to have all the details not only for the incoming channel
 but for the outgoing one as well. Furthermore, one needs to be able to 
 tweak the B-leg CDRs like he does with the incoming legs. So what
 needs to be done in my opinion is record every B-leg CDR when such an 
 event occurs and give the user access to the CDR info by
 extending the CDR() function (so that one can specify the channel of the 
 CDR that is being tweaked) or creating a seperate one for
 the outgoing channels.

 Grey Man wrote:
   
 
   
 
 I've taken the liberty of starting a new thread to discuss the design
 of the Asterisk CDR mechanism. The discussion has been kindly
 initiated by murf putting together a proposal (link ommitted to see if
 email gets accepted).

 After reading the proposal I still don't think it's the right way to
 go. To my mind adding more channel variables increases the complexity
 in a situation that is already overly so. I think it's a mistake to
 try and think about all the different call scenarios and come up with
 little tricks for the more complicated ones. There will always be
 something missed; app_shotgun initiates calls to 100 random numbers
 and as soon as three or more calls are answered it will start randonly
 transferring them amongst each other at 2 second intervals.

 I think it's important to clarify at the outset what a CDR should be.
 The most fundamental requirement for CDRs is that they accurately
 record the following pieces of information for EVERY call entering or
 leaving the system (note every means every and not; channel calls
 but not peer calls).

 1. Destination (aka as A Number)
 2. 

Re: [asterisk-users] Disabling Call-Waiting

[Sebastian]

But in this case I'm using queue not dial.

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yehavi
Bourvine
Sent: martes, 25 de noviembre de 2008 05:33 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disabling Call-Waiting

 

Try looking at the DEV_STATE function (available separately on
Asterisk-1.4). It will tell you the status of the phone before you call the
Dial() application.

 

  __Yehavi:

2008/11/25 Sebastian [EMAIL PROTECTED]

If is deprecated how do you treat a queue (realtime), that has to have just
one call for agent??

Thanks




-Original Message-
From: [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: martes, 25 de noviembre de 2008 03:37 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disabling Call-Waiting

On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
 Thanks for the responses.  I'll look into the phone devices themselves.

 I am wondering, if call-limit did not solve my problem, what is the
 call-limit parameter supposed to do anyway?

The call-limit is actually kind of deprecated.  Using the GROUP() function
is now the preferred way to do this.

exten = 1234,1,Set(GROUP()=foo)
exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
exten = 1234,n,DIal(SIP/tilghman,30)
exten = 1234,n(voicemail),Voicemail(1234,
${IF($[${DIALSTATUS}=BUSY]?b:u)})

--
Tilghman

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The message was checked by ESET Smart Security.

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Re: [asterisk-users] pick up IAX2 calls

[Tim Panton]

I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in  mixture.

I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8

So if you have
Dial(IAX/fredSIP/billzap/mark)
then someone in the same group as fred can pickup with IAX
and someone in the same group as bill can pickup with SIP
etc.

So it's an asterisk thing, not an IAX thing per-se.

Tim.

P.S.
(you could try putting in a dummy 'fred' entry into Dial and iax.conf.)

T.

On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote:


hi

thanks Luis , but doesn't work.
For SIP extensions works well *8, but for IAX a tried *8 and ** +  
iax extension and didn't works


Luis Morales wrote:


Try with ** + iax extension

Regards,

Luis Morales

On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
[EMAIL PROTECTED] wrote:


Hi

Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2  
extensions.

Any idea?

thanks



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[asterisk-users] half channel audio after upgrade to 1.4.18

[Jerry Geis]

I upgraded from 1.2 to 1.4.18

After upgrading I get half channel audio on SOME phones.

I have Cisco 7960 that works, I have a wireless polycom 8002 phone that 
works.
However, my polycom 501's are getting half channel audio on EXTERNAL calls.
Internal calls are OK.

I have enabled nat=yes on all phones.

What is something else I can try? Any thoughts on why half channel audio 
after the upgrade?

Thanks,

Jerry

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Re: [asterisk-users] Disabling Call-Waiting

[Elliot Murdock]

Hello,

I am wondering if a queue feature that blocks call-waiting should be
submitted.  As opposed to a regular line, where a call-waiting feature makes
sense, for a queue, call-waiting just doesn't.  Most queue agents are taking
phone calls in a strict order and accordingly, an annoying call-waiting beep
doesn't really make sense at all for agents to hear.

Elliot

On Tue, Nov 25, 2008 at 10:40 PM, Sebastian [EMAIL PROTECTED] wrote:

  But in this case I'm using queue not dial…





 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Yehavi Bourvine
 *Sent:* martes, 25 de noviembre de 2008 05:33 p.m.

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Disabling Call-Waiting



 Try looking at the DEV_STATE function (available separately on
 Asterisk-1.4). It will tell you the status of the phone before you call the
 Dial() application.



   __Yehavi:

 2008/11/25 Sebastian [EMAIL PROTECTED]

 If is deprecated how do you treat a queue (realtime), that has to have just
 one call for agent??

 Thanks




 -Original Message-
 From: [EMAIL PROTECTED]

 [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
 Lesher
 Sent: martes, 25 de noviembre de 2008 03:37 p.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Disabling Call-Waiting

 On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
  Thanks for the responses.  I'll look into the phone devices themselves.
 
  I am wondering, if call-limit did not solve my problem, what is the
  call-limit parameter supposed to do anyway?

 The call-limit is actually kind of deprecated.  Using the GROUP() function
 is now the preferred way to do this.

 exten = 1234,1,Set(GROUP()=foo)
 exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
 exten = 1234,n,DIal(SIP/tilghman,30)
 exten = 1234,n(voicemail),Voicemail(1234,
 ${IF($[${DIALSTATUS}=BUSY]?b:u)})

 --
 Tilghman

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 __ Information from ESET Smart Security, version of virus signature
 database 2911 (20080229) __

 The message was checked by ESET Smart Security.

 http://www.eset.com



 __ Information from ESET Smart Security, version of virus signature
 database 2911 (20080229) __

 The message was checked by ESET Smart Security.

 http://www.eset.com



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Re: [asterisk-users] two sip listening ports for single asterisk

[Rizwan Hisham]

Hi guys,
I told my network admin to do what was advised in this thread. It works very
well for incoming calls but outgoing calls hangup exactly after 20 secs
everytime while displaying the following message on cli:

v[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1910 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for seqno
102 (Critical Response)
[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1927 retrans_pkt: Hanging up
call [EMAIL PROTECTED] - no reply to our critical packet.
  == Spawn extension (macro-rating, s, 104) exited non-zero on
'SIP/saad-a8b83300' in macro 'rating'
  == Spawn extension (macro-rating, s, 104) exited non-zero on
'SIP/saad-a8b83300'

Also, this happens only with certain network conditions. in my office the
outgoing call hangsup everytime but when i dial from home, both incoming and
outgoing calls are fine. So i guess the problem is with user network
configuration. asterisk for other users who are using different port to
register is listening without any problems.


here is my network scenario(from where i make call):
we have a zyxel dsl modem connected to our ISP line
then we have a D-Link switch connected to the dsl modem
then we have sipura 2100 connected to that switch
IP addresses are in the range of 192.168.0.0
NAT type on router = SUA ONLY

On Thu, Nov 20, 2008 at 5:16 PM, Matthew J. Roth [EMAIL PROTECTED] wrote:

 Mike wrote:
  I tried using this iptables sample, and did not see duplicate packets
  on '--to-ports' port
 
  Has some verified this is working for them?
 
  I listened on both ports with tcpdump command.

 Mike,

 I can confirm that it's working.  Admittedly, I never looked at the
 packets with tcpdump because this *just worked* for me.  Calls that were
 sent to both ports (5060 and 5062) made it to Asterisk which was only
 listening on port 5060.  What's your experience with actual calls?

 As the original poster, I understand if you want third-party
 verification.  I *thought* this was a slamdunk but I'm not an iptables
 guru so I'd like it, too.

 What does the output of iptables-save and lsmod look like?  Here's
 mine, trimmed for relevancy:

 [EMAIL PROTECTED] ~]# iptables-save
 # Generated by iptables-save v1.3.5 on Thu Nov 20 12:03:21 2008
 *nat
 :PREROUTING ACCEPT [5579:1727747]
 :POSTROUTING ACCEPT [1943:176116]
 :OUTPUT ACCEPT [1943:176116]
 -A PREROUTING -i eth2 -p udp -m udp --dport 5062 -j REDIRECT --to-ports
 5060
 COMMIT
 # Completed on Thu Nov 20 12:03:21 2008

 [EMAIL PROTECTED] ~]# lsmod
 Module  Size  Used by
 ip_conntrack_netbios_ns36033  0
 ipt_REDIRECT   35009  1
 xt_tcpudp  36417  1
 iptable_nat40773  1
 ip_nat 53101  2 ipt_REDIRECT,iptable_nat
 ip_conntrack   91237  3
 ip_conntrack_netbios_ns,iptable_nat,ip_nat
 nfnetlink  40457  2 ip_nat,ip_conntrack
 ip_tables  55329  1 iptable_nat
 x_tables   50377  4
 ipt_REDIRECT,xt_tcpudp,iptable_nat,ip_tables

 Regards,

 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer


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-- 
Best Regards
Rizwan Hisham
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Re: [asterisk-users] CDR Desgin

[Benny Amorsen]

Steve Murphy [EMAIL PROTECTED] writes:

 I'll modify my RFC to reflect this line of thinking. Yes, it is a
 bit of shift. And, yes, the implementation will make this new
 approach optional, and not default.

I believe the whole approach is sound and I just want to voice my
support for it.


/Benny


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Re: [asterisk-users] Disabling Call-Waiting

[Benny Amorsen]

Elliot Murdock [EMAIL PROTECTED] writes:

 I am wondering if a queue feature that blocks call-waiting should be
 submitted.

Doesn't Queue() already disregard busy phones? I must admit that we
run with callwaiting turned off, so it isn't something I get to test
very often, but I could have sworn that Queue() didn't send useless
calls to phones it knows are busy.


/Benny


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[asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

Hello, everybody!

I need help connecting my Cisco AS5350 to Asterisk.

What i want to do is forward all outgoing calls from Asterisk server to
Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.

How could this be done?

Thanks in advance

Atif Shahzad
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[asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

Hello, everybody!

I need help connecting my Cisco AS5350 to Asterisk.

What i want to do is forward all outgoing calls from Asterisk server to
Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.

How could this be done?

Thanks in advance

Atif Shahzad
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[asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

Hello, everybody!

I need help connecting my Cisco AS5350 to Asterisk.

What i want to do is forward all outgoing calls from Asterisk server to
Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.

How could this be done?

Thanks in advance

Atif Shahzad
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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind 
that the gateway shunts calls POTS-VOIP and VOIP-POTS by default, so 
you can use the same destination pattern matching for both in this 
simple scenario, but if it gets any more complicated than that, some 
degree of translation is almost certainly required.

The process can be fairly complex, but the general idea, if you have 
your TDM side set up, is:

dial-peer voice 500 voip
  description Asterisk
  destination-pattern .T
  progress_ind setup enable 3
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  dtmf-relay rtp-nte
  no vad

dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10


A T I F wrote:

 Hello, everybody!
 
 I need help connecting my Cisco AS5350 to Asterisk.
 
 What i want to do is forward all outgoing calls from Asterisk server to 
 Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.
 
 How could this be done?
 
 Thanks in advance
 
 Atif Shahzad
 
 
 
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Disabling Call-Waiting

[Paul Hales]

Benny Amorsen wrote:
 Elliot Murdock [EMAIL PROTECTED] writes:

   
 I am wondering if a queue feature that blocks call-waiting should be
 submitted.
 

 Doesn't Queue() already disregard busy phones? I must admit that we
 run with callwaiting turned off, so it isn't something I get to test
 very often, but I could have sworn that Queue() didn't send useless
 calls to phones it knows are busy.


 /Benny

   
There's a 'ringinuse' field for queues that you can set to
enable/disable this function (calling phones that are in use) so you can
make the decision yourself, rather than have Asterisk make an arbitrary
decision for you.

later,

PaulH

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Re: [asterisk-users] half channel audio after upgrade to 1.4.18

[Paul Hales]

Jerry Geis wrote:
 I upgraded from 1.2 to 1.4.18

 After upgrading I get half channel audio on SOME phones.

 I have Cisco 7960 that works, I have a wireless polycom 8002 phone that 
 works.
 However, my polycom 501's are getting half channel audio on EXTERNAL calls.
 Internal calls are OK.

 I have enabled nat=yes on all phones.

 What is something else I can try? Any thoughts on why half channel audio 
 after the upgrade?

   
'canreinvite=no' can also help with this.

PaulH

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

The procedure you explain is for outbound or inbound for Asterisk or Can you
tell me the procedure for only outbound from my Asterisk server to Cisco
5350?

On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov [EMAIL PROTECTED]wrote:

 Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind
 that the gateway shunts calls POTS-VOIP and VOIP-POTS by default, so
 you can use the same destination pattern matching for both in this
 simple scenario, but if it gets any more complicated than that, some
 degree of translation is almost certainly required.

 The process can be fairly complex, but the general idea, if you have
 your TDM side set up, is:

 dial-peer voice 500 voip
  description Asterisk
  destination-pattern .T
  progress_ind setup enable 3
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  dtmf-relay rtp-nte
  no vad

 dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10


 A T I F wrote:

  Hello, everybody!
 
  I need help connecting my Cisco AS5350 to Asterisk.
 
  What i want to do is forward all outgoing calls from Asterisk server to
  Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.
 
  How could this be done?
 
  Thanks in advance
 
  Atif Shahzad
 
 
  
 
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 Evariste Systems
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 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

Alex,

1 more thing my gateway is configured with H.323 so tell me how can I
configure it with SIP?

On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov [EMAIL PROTECTED]wrote:

 Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind
 that the gateway shunts calls POTS-VOIP and VOIP-POTS by default, so
 you can use the same destination pattern matching for both in this
 simple scenario, but if it gets any more complicated than that, some
 degree of translation is almost certainly required.

 The process can be fairly complex, but the general idea, if you have
 your TDM side set up, is:

 dial-peer voice 500 voip
  description Asterisk
  destination-pattern .T
  progress_ind setup enable 3
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  dtmf-relay rtp-nte
  no vad

 dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10


 A T I F wrote:

  Hello, everybody!
 
  I need help connecting my Cisco AS5350 to Asterisk.
 
  What i want to do is forward all outgoing calls from Asterisk server to
  Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.
 
  How could this be done?
 
  Thanks in advance
 
  Atif Shahzad
 
 
  
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users


 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

Use only the second dial peer.

A T I F wrote:

 The procedure you explain is for outbound or inbound for Asterisk or Can 
 you tell me the procedure for only outbound from my Asterisk server to 
 Cisco 5350?
 
 On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
 Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind
 that the gateway shunts calls POTS-VOIP and VOIP-POTS by default, so
 you can use the same destination pattern matching for both in this
 simple scenario, but if it gets any more complicated than that, some
 degree of translation is almost certainly required.
 
 The process can be fairly complex, but the general idea, if you have
 your TDM side set up, is:
 
 dial-peer voice 500 voip
  description Asterisk
  destination-pattern .T
  progress_ind setup enable 3
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  dtmf-relay rtp-nte
  no vad
 
 dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10
 
 
 A T I F wrote:
 
   Hello, everybody!
  
   I need help connecting my Cisco AS5350 to Asterisk.
  
   What i want to do is forward all outgoing calls from Asterisk
 server to
   Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.
  
   How could this be done?
  
   Thanks in advance
  
   Atif Shahzad
  
  
  
 
  
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  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599
 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

My attention to my dial peer.  It has nothing about H.323 and much about 
SIP.

A T I F wrote:

 Alex,
 
 1 more thing my gateway is configured with H.323 so tell me how can I 
 configure it with SIP?
 
 On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
 Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind
 that the gateway shunts calls POTS-VOIP and VOIP-POTS by default, so
 you can use the same destination pattern matching for both in this
 simple scenario, but if it gets any more complicated than that, some
 degree of translation is almost certainly required.
 
 The process can be fairly complex, but the general idea, if you have
 your TDM side set up, is:
 
 dial-peer voice 500 voip
  description Asterisk
  destination-pattern .T
  progress_ind setup enable 3
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  dtmf-relay rtp-nte
  no vad
 
 dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10
 
 
 A T I F wrote:
 
   Hello, everybody!
  
   I need help connecting my Cisco AS5350 to Asterisk.
  
   What i want to do is forward all outgoing calls from Asterisk
 server to
   Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP.
  
   How could this be done?
  
   Thanks in advance
  
   Atif Shahzad
  
  
  
 
  
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   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599
 
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-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

1.   dial-peer voice 500 voip

I use this configuration for inbound to asterisk.

2. dial-peer voice 510 pots
description Fancy PRI - Outgoing
huntstop
destination-pattern .T
direct-inward-dial
forward-digits 10

And use this configuration for outbound from asterisk to Cisco 5350 right?

On Tue, Nov 25, 2008 at 3:42 PM, Alex Balashov [EMAIL PROTECTED]wrote:

 My attention to my dial peer.  It has nothing about H.323 and much about
 SIP.

 A T I F wrote:

  Alex,
 
  1 more thing my gateway is configured with H.323 so tell me how can I
  configure it with SIP?
 
  On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
  Set up a SIP dial peer and an outbound POTS dial peer.  Bear in mind
  that the gateway shunts calls POTS-VOIP and VOIP-POTS by default,
 so
  you can use the same destination pattern matching for both in this
  simple scenario, but if it gets any more complicated than that, some
  degree of translation is almost certainly required.
 
  The process can be fairly complex, but the general idea, if you have
  your TDM side set up, is:
 
  dial-peer voice 500 voip
   description Asterisk
   destination-pattern .T
   progress_ind setup enable 3
   voice-class codec 1
   session protocol sipv2
   session target ipv4:ip.addr.of.asterisk
   session transport udp
   dtmf-relay rtp-nte
   no vad
 
  dial-peer voice 510 pots
   description Fancy PRI - Outgoing
   huntstop
   destination-pattern .T
   direct-inward-dial
   forward-digits 10
 
 
  A T I F wrote:
 
Hello, everybody!
   
I need help connecting my Cisco AS5350 to Asterisk.
   
What i want to do is forward all outgoing calls from Asterisk
  server to
Cisco AS5350, and from Cisco 5350 to my Asterisk server, using
 SIP.
   
How could this be done?
   
Thanks in advance
   
Atif Shahzad
   
   
   
 
 
   
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 http://www.api-digital.com --
   
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  --
  Alex Balashov
  Evariste Systems
  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (706) 338-8599
 
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http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  
 
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 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

A T I F wrote:
 1.   dial-peer voice 500 voip
 
 I use this configuration for inbound to asterisk.
 
 2. dial-peer voice 510 pots
 description Fancy PRI - Outgoing
 huntstop
 destination-pattern .T
 direct-inward-dial
 forward-digits 10
 
 And use this configuration for outbound from asterisk to Cisco 5350 right?

Yep.

You may wish to have an incoming peer on the VoIP side to match first to 
do various translations in the future.  It's generally considered better 
form.  Then the call will enter in this dial peer and exit in 510.

dial-peer voice 801 voip
  description Asterisk - inbound
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  incoming called-number .T
  dtmf-relay rtp-nte
  no vad


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] CDR Desgin

[Freddi Hansen]

 To me the obvious answer is to provide a CDR for every call leg so for
  A calling B via Asterisk there would be two CDRs produced. It's far
  far easier to disregard the unwanted CDRs than it is to try and
  generate the missing ones and in some cases it's virtually impossible.
  If it's weighed up I think people would vote to have accurate CDRs
  ahead of anything else and if single legs are the best way to do that
  then it's the way it should be done.
  
  In addition with single leg CDRs it will solve the dilemna about
  acommodating every possible call scenario that I know has caused you a
  lot of consternation over the last 18 months.
  
  Sure it's a change from the current situation so maybe needs to use
  the standard apporach of a configuration setting to opt in initially
  before becoming the default in the subsequent major release.
   


 OK, Greyman, your logic is solid. If we provide a CDR implementation
 that just generates the individual call legs, and ties them together via
 the linkedid
 (see team/group/newcdr), then both camps should be able to derive the
 info
 they need for billing, via hopefully not-overly-complex SQL queries to a
 backend db.

 I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of
 shift.
 And, yes, the implementation will make this new approach optional, and
 not
 default. But, pardon if I make it available via the CEL domain come
 implementation time.


 It should take me a week to rehash my document; perhaps longer (I'm in
 bugfix mode, and this borderline development work); in the meantime,
 those with decided CDR needs might make their wishes known, if they do
 not think this approach will work. Speak now, or forever hold your
 peace; or forever complain... or whatever.
 If this is particularly distressing to you, perhaps your fears might be
 slightly assuaged when you read the details...
   
I was part of a team that did design a multiservice billing system about 
15 years ago and the solution people seems to agree on here (and me to) 
looks pretty much the same i.e one call may consist of several calls 
legs. In addition to the linkedid it would be nice to have an indication 
in the cdr that tells us that 'this is the lastone on this  linked id'.
Our experience was that  we shouldn't  for load reasons work with cdr's 
in the immidiate multileg format in the DB. So we did collect cdr's in a 
tmp DB until we got the the record with end marker set. We would then 
produce our final cdr for the actual service, store it in billing col. 
and delete it from the multileg col. When a new service is created we 
only have to make a the new customized cdr, we don't have to touch the 
generation of the multileg format.  

Freddi




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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

Alex,

I am new to *5350* my senerio is this;

*1. ASTERISK  ---outgoing--CISCO5350 (both have live IP configured)

2. ASTERISK -incomingCISCO5350*

I need only configurations for Cisco for both in coming n outgoing to
asterisk. IF you need configuration of my Cisco Gateway I will provide you.
Sorry to bother you again.  I have to make up assignment on it hope you help
me out.

Atif Shahzad.

On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov [EMAIL PROTECTED]wrote:

 A T I F wrote:
  1.   dial-peer voice 500 voip
 
  I use this configuration for inbound to asterisk.
 
  2. dial-peer voice 510 pots
  description Fancy PRI - Outgoing
  huntstop
  destination-pattern .T
  direct-inward-dial
  forward-digits 10
 
  And use this configuration for outbound from asterisk to Cisco 5350
 right?

 Yep.

 You may wish to have an incoming peer on the VoIP side to match first to
 do various translations in the future.  It's generally considered better
 form.  Then the call will enter in this dial peer and exit in 510.

 dial-peer voice 801 voip
  description Asterisk - inbound
   voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
   incoming called-number .T
   dtmf-relay rtp-nte
  no vad


 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

This is not the homework list.

A T I F wrote:

 Alex,
 
 I am new to _*5350*_ my senerio is this;
 
 *1. ASTERISK  ---outgoing--CISCO5350 (both have live IP configured)
 
 2. ASTERISK -incomingCISCO5350*
 
 I need only configurations for Cisco for both in coming n outgoing to 
 asterisk. IF you need configuration of my Cisco Gateway I will provide 
 you. Sorry to bother you again.  I have to make up assignment on it hope 
 you help me out.
 
 Atif Shahzad.
 
 On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
 A T I F wrote:
   1.   dial-peer voice 500 voip
  
   I use this configuration for inbound to asterisk.
  
   2. dial-peer voice 510 pots
   description Fancy PRI - Outgoing
   huntstop
   destination-pattern .T
   direct-inward-dial
   forward-digits 10
  
   And use this configuration for outbound from asterisk to Cisco
 5350 right?
 
 Yep.
 
 You may wish to have an incoming peer on the VoIP side to match first to
 do various translations in the future.  It's generally considered better
 form.  Then the call will enter in this dial peer and exit in 510.
 
 dial-peer voice 801 voip
  description Asterisk - inbound
  voice-class codec 1
  session protocol sipv2
  session target ipv4:ip.addr.of.asterisk
  session transport udp
  incoming called-number .T
  dtmf-relay rtp-nte
  no vad
 
 
 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599
 
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
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   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
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http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[A T I F]

I mean to say that I have assigned this task from my management to configure
it, I am familiar with Asterisk but first time I am using Cisco 5350.



On Tue, Nov 25, 2008 at 4:19 PM, Alex Balashov [EMAIL PROTECTED]wrote:

 This is not the homework list.

 A T I F wrote:

  Alex,
 
  I am new to _*5350*_ my senerio is this;
 
  *1. ASTERISK  ---outgoing--CISCO5350 (both have live IP configured)
 
  2. ASTERISK -incomingCISCO5350*
 
  I need only configurations for Cisco for both in coming n outgoing to
  asterisk. IF you need configuration of my Cisco Gateway I will provide
  you. Sorry to bother you again.  I have to make up assignment on it hope
  you help me out.
 
  Atif Shahzad.
 
  On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
  A T I F wrote:
1.   dial-peer voice 500 voip
   
I use this configuration for inbound to asterisk.
   
2. dial-peer voice 510 pots
description Fancy PRI - Outgoing
huntstop
destination-pattern .T
direct-inward-dial
forward-digits 10
   
And use this configuration for outbound from asterisk to Cisco
  5350 right?
 
  Yep.
 
  You may wish to have an incoming peer on the VoIP side to match first
 to
  do various translations in the future.  It's generally considered
 better
  form.  Then the call will enter in this dial peer and exit in 510.
 
  dial-peer voice 801 voip
   description Asterisk - inbound
   voice-class codec 1
   session protocol sipv2
   session target ipv4:ip.addr.of.asterisk
   session transport udp
   incoming called-number .T
   dtmf-relay rtp-nte
   no vad
 
 
  --
  Alex Balashov
  Evariste Systems
  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (706) 338-8599
 
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http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  
 
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 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Connecting AS5350XM with Asterisk

[Alex Balashov]

I provided you all the information you need in an elementary sense.

Configuring the device comprehensively is a rather lengthy subject and 
cannot be meaningfully addressed in the scope of an email thread.  The 
answer basically boils down to, learn the Cisco VFC platform.

There are plenty of sample configurations that can be found online at:

http://www.letmegooglethatforyou.com/?q=Cisco+AS5300+asterisk

A T I F wrote:

 I mean to say that I have assigned this task from my management to 
 configure it, I am familiar with Asterisk but first time I am using 
 Cisco 5350.
 
 
 
 On Tue, Nov 25, 2008 at 4:19 PM, Alex Balashov 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
 This is not the homework list.
 
 A T I F wrote:
 
   Alex,
  
   I am new to _*5350*_ my senerio is this;
  
   *1. ASTERISK  ---outgoing--CISCO5350 (both have live IP
 configured)
  
   2. ASTERISK -incomingCISCO5350*
  
   I need only configurations for Cisco for both in coming n outgoing to
   asterisk. IF you need configuration of my Cisco Gateway I will
 provide
   you. Sorry to bother you again.  I have to make up assignment on
 it hope
   you help me out.
  
   Atif Shahzad.
  
   On Tue, Nov 25, 2008 at 3:58 PM, Alex Balashov
   [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
  
   A T I F wrote:
 1.   dial-peer voice 500 voip

 I use this configuration for inbound to asterisk.

 2. dial-peer voice 510 pots
 description Fancy PRI - Outgoing
 huntstop
 destination-pattern .T
 direct-inward-dial
 forward-digits 10

 And use this configuration for outbound from asterisk to Cisco
   5350 right?
  
   Yep.
  
   You may wish to have an incoming peer on the VoIP side to
 match first to
   do various translations in the future.  It's generally
 considered better
   form.  Then the call will enter in this dial peer and exit in
 510.
  
   dial-peer voice 801 voip
description Asterisk - inbound
voice-class codec 1
session protocol sipv2
session target ipv4:ip.addr.of.asterisk
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
no vad
  
  
   --
   Alex Balashov
   Evariste Systems
   Web: http://www.evaristesys.com/
   Tel: (+1) (678) 954-0670
   Direct : (+1) (678) 954-0671
   Mobile : (+1) (706) 338-8599
  
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 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] pick up IAX2 calls

[Bruno Castelo Branco]

hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 
for all IAX extensions in iax.conf. Didn't works for while.

thanks

Tim Panton wrote:

I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in  mixture.

I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8

So if you have 
Dial(IAX/fredSIP/billzap/mark)

then someone in the same group as fred can pickup with IAX
and someone in the same group as bill can pickup with SIP
etc.

So it's an asterisk thing, not an IAX thing per-se.

Tim.

P.S.
(you could try putting in a dummy 'fred' entry into Dial and iax.conf.) 


T.

On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote:


hi

thanks Luis , but doesn't work.
For SIP extensions works well *8, but for IAX a tried *8 and ** + iax 
extension and didn't works


Luis Morales wrote:

Try with ** + iax extension

Regards,

Luis Morales

On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
  

Hi

Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2 extensions.
Any idea?

thanks



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[asterisk-users] bridging - Didn't get a frame from channel

[Tony Gaspar]

Hi,
 
 
I am having a difficulty with 
getting two realtime user’s to bridge on answer. I have managed successfully to 
bridge the same two users/channels via the Bridge Manager api command and 
confirm that the two communicate directly bypassing the asterisk server (I 
confirmed this with Wireshark). 
 
 
Does anyone have some ideas? I have 
put some log entries below. 
 
I haven’t attached my dialplan. Some 
behaviours I have discovered are: 1) If user A dials directly to the call 
number 
of user B, bridging works on answer and the failed log entries below don’t 
occur. 2) If user A dials into a queue and waits for User B (who is registered 
in the queue). Then when User B answers, the bridging failures occur as in the 
log entries below.
 
Thank 
you.
 
Tony 
Gaspar
 
 
 
 
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Internal timing is disabled (option_internal_timing=0 
chan-timingfd=26)
[Nov 25 17:47:38] DEBUG[27462] 
devicestate.c: Notification of state change to be queued on device/channel 
IAX2/asxop
[Nov 25 17:47:38] VERBOSE[27462] 
logger.c: -- IAX2/asxop-14202 answered 
IAX2/wally-10884
[Nov 25 17:47:38] DEBUG[25220] 
devicestate.c: No provider found, checking channel drivers for IAX2 - 
asxop
[Nov 25 17:47:38] DEBUG[25220] 
chan_iax2.c: Checking device state for device asxop
[Nov 25 17:47:38] DEBUG[25220] 
chan_iax2.c: iax2_devicestate: Found peer. What's device state of asxop? 
addr=984047307, defaddr=0 maxms=5000, lastms=191
[Nov 25 17:47:38] DEBUG[25220] 
devicestate.c: Changing state for IAX2/asxop - state 2 (In 
use)
[Nov 25 17:47:38] DEBUG[25225] 
app_queue.c: Device 'IAX2/asxop' changed to state '2' (In 
use)
[Nov 25 17:47:38] DEBUG[27462] 
app_queue.c: Next is 'IAX2/mark' with metric 1
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Set channel IAX2/wally-10884 to write format 
ilbc
[Nov 25 17:47:38] VERBOSE[27462] 
logger.c: -- Stopped music on hold on 
IAX2/wally-10884
[Nov 25 17:47:38] DEBUG[27462] 
channel.c: Scheduling timer at 0 sample intervals
[Nov 25 17:47:38] DEBUG[27462] 
app_queue.c: Starting MixMonitor as requested.
[Nov 25 17:47:38] DEBUG[27462] 
app_queue.c: Arguments being passed to MixMonitor: 
1227656841.388.wav,b
[Nov 25 17:47:38] DEBUG[27462] 
app_queue.c: Queue 'assist' Leave, Channel 
'IAX2/wally-10884'
[Nov 25 17:47:38] VERBOSE[27464] 
logger.c:   == Begin MixMonitor Recording 
IAX2/wally-10884
[Nov 25 17:47:38] DEBUG[27462] 
features.c: bridge answer set, chan answer set
[Nov 25 17:47:38] DEBUG[27464] 
audiohook.c: Failed to get 160 samples from write factory 
0x7f5aa4095fa0
[Nov 25 17:47:38] DEBUG[27464] 
audiohook.c: Failed to get 160 samples from write factory 
0x7f5aa4095fa0
[Nov 

[asterisk-users] Ring/Off-hook in strange state 6 channel X

[research]

Greetings List

I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all
of them give me the error Ring/Off-hook in strange state 6.

Whenever the caller hangup, the call continue to execute until it hits the
hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=no
but problem still persist. I also tried to use different PABX in vain. GSM
modem (FUSION100) also produces no useful result

Please help

Sam Muro


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Re: [asterisk-users] The sound is played but I did not hear

[jhon digital21]

same result [?]

2008/11/25 Doug Lytle [EMAIL PROTECTED]

 Try putting a Wait(1) before playing the sound effect.

 Doug

 --

 Ben Franklin quote:

 Those who would give up Essential Liberty to purchase a little Temporary
 Safety, deserve neither Liberty nor Safety.


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