Re: [asterisk-users] VoIP monitoring tools

[Neeraj Chand]

Homer for voip / flow capture


Smoke ping has a sip based server test feature in it as well

Sent from my iPhone

> On 27 Sep. 2016, at 7:17 pm, "sysad...@reed-media.com" 
>  wrote:
> 
> Hello,
> 
> you can have a look on Homer
> 
> http://sipcapture.org/
> 
> regards
> 
> 
> 
>> On 27/09/2016 10:39, Gholamreza Sabery wrote:
>> Hello,
>> 
>> For service monitoring you can use tools like sipsak in combination with 
>> Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the health of 
>> your servers. This way you have both top-down and bottom-up monitoring. For 
>> monitoring call quality you can use tools like VoIP Monitor (it is not free).
>> 
>> Regards
>> 
>> 
>>> On Tue, Sep 27, 2016 at 12:03 PM, Nitesh Bansal  
>>> wrote:
>>> Hello all,
>>> 
>>> The question isn't directly related to Asterisk, but I'm looking for 
>>> recommendations
>>> for a monitoring tool to monitor the health of Asterisk instances running 
>>> in Production.
>>> 
>>> Ideally, the tool should be able to generate   monitoring 
>>> traffic (OPTIONS ping or INVITE),
>>> use the response/no response from Asterisk to store the health of an 
>>> Asterisk instance running 
>>> somewhere in the DB.
>>> 
>>> Thanks,
>>> Nitesh Bansal
>>> 
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>>> 
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>> 
>> 
>> 
> 
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Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

[Neeraj Chand]

How about running a second asterisk instance on the same box with different 
IP/Port combo
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Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

[Neeraj Chand]

And if we were the trunk provider in this case - I would probably trunk it to a 
separate machine, or run a second instance of asterisk on the same machine and 
bind it to the secondary IP address / sip port combo, peer the two instances 
and then send it out via the second instance. 



Sent from my iPhone

> On 26 May 2016, at 7:46 AM, Glenn Geller (VDOPh)  wrote:
> 
> Hi,
> 
> Usually, the trunk provider(s) provide a mechanism to support this, and it's 
> the "Tech Prefix" or just "Prefix".
> 
> So, for Tenant 1, send 12348005551212 and for Tenant 2, send 56788005551212.
> 
> They'll then strip the prefix, and send along to 8005551212
> 
> Most trunk providers worth anything will support this type of termination.
> 
> Hope this helps,
> 
> Glenn @ VDO
> 
>> On Wed, May 25, 2016 at 2:13 PM, Attila Megyeri  
>> wrote:
>> Hi!
>> 
>>  
>> 
>> I would like to reopen a discussion that I saw a couple of years ago, with 
>> the subject  “Sending Calls via SIP trunk from two different IP addresses 
>> from same Asterisk Machine”
>> 
>>  
>> 
>> The use case is simpe: There are providers that want to see a separate 
>> source IP address for each of their customers SIP trunks. Now, if we have an 
>> asterisk box with several customers, we have a problem.
>> 
>>  
>> 
>> Does anyone have experience in this topic? How could we send outgoing calls 
>> (to the same destination IP) from different source IPs depending on the 
>> caller ID (Based on From: field, sip account, preferred-identity, whatever).
>> 
>>  
>> 
>> I was thinking about some Kamailio, or SBC that would take the calls from 
>> asterisk using a user/pass authentication on a single interface, and 
>> initiate calls from a dedicated IP address for each customer.
>> 
>> Any better idea?
>> 
>>  
>> 
>> Thanks
>> 
>>  
>> 
>> 
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[asterisk-users] Hints realtime table structure Ast 11

[Neeraj Chand]

Hi All,

Has anyone used hints in realtime ?

(As in storing and loading hints from odbc)

I cannot find a table structure for this anywhere...?


Thanks

Neeraj

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Re: [asterisk-users] Fwd: Unable to place outbound calls

[Neeraj Chand]

Could potentially be a NAT problem. Could you do a sip debug and see whether 
there is any traffic to / from the address?

Sent from my iPhone

> On 15 Mar 2016, at 9:52 PM, Administrator TOOTAI  wrote:
> 
> Le 15/03/2016 11:20, Feroz Ahmed a écrit :
>> Hi I need help
> 
> Hello Ahmed
> 
>> 
>> This is the error:
> 
> [...]
> 
>> [Mar 14 19:55:15] WARNING[20595][C-000b]: app_dial.c:2437
>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>> Subscriber absent)
>>   == Everyone is busy/congested at this time (1:0/0/1)
>> -- Auto fallthrough, channel 'SIP/1001-000b' status is
>> 'CHANUNAVAIL'
> 
> CHANUNAVAIL and here is why
> 
>> sonetel/feroz.sonetel 212.72.62.126D
>>  Auto (No)  Yes5060 UNREACHABLE
> 
> Unreachable, you face a network problem.
> 
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Re: [asterisk-users] Dial your phone and contact phone from within outlook?

[Neeraj Chand]

Hi Travis,

Have a look at this:

http://www.ipcom.at/en/telephony/siptapi/

I have used this in the past to do something similar, unless you have an
Exchange Enterprise setup in which case I would suggest exploring unified
messaging

Thanks,

Neeraj

On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis  wrote:

> I am wondering what the best solution is for initiating a call from
> Outlook Contacts. I imagine something that would start the call from the
> outlook card (or similar) and then dial the user’s extension and the
> contact’s phone number and place them in a bridge.
>
>
>
> Anyone use something like this?
>
>
>
> Travis Ryan
> Director of Information Technologies
> Oscar Winski Company
> 2407 North Ninth Street
> Lafayette, IN 47905
> ry...@oscarwinski.com
> (765) 742-1102
>
>
> *We're not the IT departmentWe're the I-TEAM department!*
>
>
>
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[asterisk-users] Cisco BLF c7975 notifications not working with asterisk realtime

[Neeraj Chand]

Hi All,

I've set up asterisk 11.20.0 with the blf patch and registered cisco phones
to the server.
I can see the subscriptions and the last state as idle, type = pidf+xml

In hints I also have:
Location  Hints   DeviceState
PresenceState   Watchers
XXX@YYY Custom:XXX IdleNot_Set
2


However when the phone rings the phone statuses are not sent out to the
subscribers.

The cisco phones are configured with:


21
XXX
XXX
1


Any pointers? Tearing out whats left of a once full head of hair here.

:)

Thanks,
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Re: [asterisk-users] Pops clicks at the end of sound files

[Neeraj Chand]

Hi all, 

We recently decided to get a professionally recorded set of prompts for
our asterisk based IVRs and received these as the following: 

Bit Rate: 1536Kbps
Sample Size: 16bit
Channels: Stereo
Sample Rate: 48kHz
Format: PCM

I use Wavepad to convert it to:
Bit Rate:64Kbps
Sample Size: 8bit
Channels: Mono
Sample Rate: 48kHZ
Format: CCIT-ALAW

I copied these files to an asterisk server and then used asterisk -rx to
convert the files to g729. 

The problem I have is that at the end of every file there is a pop /
distortion after playback. 

Anyone have the same issue before? 



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Re: [asterisk-users] Features.conf - Blind Transfer

[Neeraj Chand]

Hi guys, 

I'm trying to get blind transfer to work and automatically transfer call
to another number on key sequence press. 

Extensions.conf_snippet

[from-pstn]
exten = _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer)
exten = _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1)
exten = _0399377744,n,dial(SIP/0399377704@c5400-02,T)



[to-pstn]
Exten = _XXX.,1,dial(sip/0388924326@ c5400-01)


Features.conf_snippet

[featuremap]
blindxfer = #1 


on #1 all I get is silence, and debug shows call going to 'i'

ast ver 1.4.24

Appreciate your help

Thanks, 





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Re: [asterisk-users] record individual callers in confbridge

[Neeraj Chand]

 I'd suggest try recording in ulaw first and then convert all to wav after.

 It may have something to do with timing since you're using iax, what are you 
 using as a timing source? Hardware or software?

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[asterisk-users] record individual callers in confbridge

[Neeraj Chand]

While you're testing, capture iax2 debug info as well as that may point to 
other factors/issues.


Possible fixes:

File conversion:
Yes asterisk can do the conversion

File convert xyz.ulaw xyz.wav

As long as you selected wav format in the initial build.

Timing module for iax:

Check your modules directory to see which timing sources you have installed. 
Res_pthread, res_timerfd and dahdi_dummy would be located there.

Move res_pthread and res_timerfd(if it exists) out of the modules directory and 
restart * to force dahdi dummy.


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[asterisk-users] Ast 1.8_CentOS5.5 with timerfd as timing source

[Neeraj Chand]

Hi All

Just finished setting up a vm with centos 5.5 and asterisk 1.8.3

Using timerfd as a timing source. 

Has anyone got a similar setup in production ? 

How's performance? 

Thanks, 

Neeraj 








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Re: [asterisk-users] Email from Dialplan

[Neeraj Chand]

I use the following: 

Exten = s,n(status-NOTIFY),System(echo '${DIALSTATUS} on
${CALLERID(num)}' at ${STRFTIME(${EPOCH},,%H%M%S)} | mail -s Call
Unsuccessful on DNIS '${ARG10}' neeraj.ch...@ocis.com.au)


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Re: [asterisk-users] MS-SQL / Freetds -- func_odbc

[Neeraj Chand]

Hi folks, 

How would I go about running a stored procedure call from asterisk via
func_odbc. 

I'm after an example entry in func_odbc if possible for ast 1.4

Also, if someone could post an insert statement that actually works,
would be nice. 

Thanks,

:) 


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Re: [asterisk-users] Asterisk ODBC Insert issue

[Neeraj Chand]

Hi guys, 

Having issues with doing an insert statement using ast 1.4.24: 

[START]
dsn=mssql-asterisk
write=INSERT INTO testdb (callarrival,callerid) VALUES
('${VAL1}','${VAL2}')


SET(ODBC_START()${TIMESTAMP},${CALLERID(num)})

No errors pop up on execute, but nothing gets inserted. 

Read and update work fine, 

Wondering where I'm going off track with this, 

Thanks, 

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Re: [asterisk-users] SKYPE - Authenticate incoming call

[Neeraj Chand]

 
 
 Hi All, 
 
 After getting licences for Skype for asterisk a while ago I finally
got
 around to setting up a server with two channels and setting up a bcp
on
 the skype end. 
 
 The idea behind this is the following: 
 
 Users can dial into the PBX, get authenticated and only after
 authentication get access to internal PBX extensions. 
 
 I CAN do this with a PIN, no sweat, but from a user perspective it
 becomes a bit clunky, i.e. password to remember, security in terms of
 pin leaks, multiple passwords for users, etc. 
 
 I was wondering if there was a way I could extract the FROM - USER
and
 assign it to a variable, then do a lookup of that username in a
database
 using ODBC to decide whether to allow or disallow access. 
 
 NOTE: The bit I need help with is extracting the FROM - USER the
rest
 of the stuff I've done already / before.  

None of this is necessary; Skype already supports restricting to calls
to only coming from users on the buddy list. So, if your PBX is
connecting to the Skype network as user 'A', and your remote users are
'B' and 'C', then *don't* setup SFA to allow calls from anyone, and
don't set it up to automatically add users to the buddy list when they
request it. Instead, manually add users B and C to A's buddy list
(using
a regular Skype client), and those are the only users that will be able
to call A.

-- 
Kevin P. Fleming

I know that already, it's a matter of convenience. 
If I go that way, then I have to manually log in to skype, and add maybe
50 / 60 users to each new user that I create [these are personal staff
accounts that wont be logged into the asterisk server via SFA, and are
not part of the group set up in BCP]
If there's something I can do on the asterisk end, then management
becomes *very* simple -- func_odbc+freetds+MS_SQL+PHP = web page to
manage users  access. 

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Re: [asterisk-users] SKYPE - Authenticate incoming call automatically

[Neeraj Chand]

Hi All, 

After getting licences for Skype for asterisk a while ago I finally got
around to setting up a server with two channels and setting up a bcp on
the skype end. 

The idea behind this is the following: 

Users can dial into the PBX, get authenticated and only after
authentication get access to internal PBX extensions. 

I CAN do this with a PIN, no sweat, but from a user perspective it
becomes a bit clunky, i.e. password to remember, security in terms of
pin leaks, multiple passwords for users, etc. 

I was wondering if there was a way I could extract the FROM - USER and
assign it to a variable, then do a lookup of that username in a database
using ODBC to decide whether to allow or disallow access. 

NOTE: The bit I need help with is extracting the FROM - USER the rest
of the stuff I've done already / before.  

Thanks, 

Neeraj. 

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Re: [asterisk-users] asterisk and cisco 2800

[Neeraj Chand]

Hi Giorgio, 

Why don't you terminate calls on the cisco router via SIP? 



--

Message: 11
Date: Fri, 02 Jul 2010 18:54:31 +0200
From: Giorgio Incantalupo gincantal...@fgasoftware.com
Subject: [asterisk-users] asterisk and cisco 2800
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c2e19c7.5090...@fgasoftware.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi all,

I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures

with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the 
cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives 
no errros, the span is up and active, green light on the card) but when 
I make a test with my iax phone, there's no way to dial the PBX and I 
get this WARNING:

[Jul  2 15:20:36] VERBOSE[15004] logger.c: -- Accepting 
AUTHENTICATED call from XXX.XXX.XXX.XXX:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
[Jul  2 15:20:36] VERBOSE[15031] logger.c: -- Executing 
[6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new 
stack
[Jul  2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of

type 'DAHDI' (cause 0 - Unknown)
[Jul  2 15:20:36] VERBOSE[15031] logger.c:   == Everyone is 
busy/congested at this time (1:0/0/1)
[Jul  2 15:20:36] VERBOSE[15031] logger.c: -- Executing 
[6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack
[Jul  2 15:20:36] VERBOSE[15031] logger.c:   == Spawn extension 
(inbound, , 2) exited non-zero on 'IAX2/1-1024'
[Jul  2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024'

Any hints?

Thank you.

Giorgio Incantalupo





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Re: [asterisk-users] CDR to MS-SQL via ODBC issue

[Neeraj Chand]

On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote:


 ---
 Message: 10
 Date: Wed, 5 May 2010 10:26:34 -0500
 From: Tilghman Lesher tles...@digium.com
 Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Message-ID: 201005051026.34929.tles...@digium.com
 Content-Type: text/plain;  charset=iso-8859-1

 On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote:
  I can connect to the database and run via isql, and also use

 func_odbc,

  etc with res_odbc configured with the same database / freetds, but I
  cannot write CDRs.
 
 Are you writing to the database with func_odbc, or just reading?  My

 gut says

 that you need to check your permissions on the database to ensure
that

 you're

 allowed to write to the CDR table.

   Hi Tilghman, yeah I thought so too at first but then, using the
 same permissions I'm doing both read  writes as well.

 On the database end, the user is setup as database_owner and has
db_read
  db_write permissions.

 I got Leif to check this with me last night, we couldn't figure it
out.

 The error that pops up is:
 cdr_odbc: Connected to asterisk-freetds-connector
 cdr_odbc: Error in PREPARE -1
 cdr_odbc: Query FAILED Call not logged!



 __

Okay, second idea is that you should very carefully examine your CDR
table
layout and ensure that the columns that you have match EXACTLY what the
module expects you to have.  If Asterisk expects you to have a column
that you
don't (or the column type is wrong), that is another reason that the
prepare
might fail.  You might consider using the cdr_adaptive_odbc driver,
instead,
as it is designed to create the insert based upon the structure of the
table.

-- 
Tilghman Lesher

-

That did the trick. Had calldate  userfield missing  I had
loguniqueid=yes.

Thanks Leif / Tilghman. 

:)

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Re: [asterisk-users] CDR to MS-SQL via ODBC issue

[Neeraj Chand]

Hi guys, 

Having issue with getting CDR to write to MS-SQL via ODBC.

cdr_odbc: Connected to freetds-connector
cdr_odbc: Error in PREPARE -1
cdr_odbc: Query FAILED Call not logged!
  == Spawn extension (cisco, ##, 2) exited non-zero on
'IAX2/ast-507


Isql test: 

[...@ asterisk]# isql freetds-connector XXX Y
+---+
| Connected!|
|   |
| sql-statement |
| help [tablename]  |
| quit  |
|   |
+---+
SQL

I can connect to the database and run via isql, and also use func_odbc,
etc with res_odbc configured with the same database / freetds, but I
cannot write CDRs. 

Any ideas would be really appreciated. 

Thanks, 

Neeraj

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Re: [asterisk-users] CDR to MS-SQL via ODBC issue

[Neeraj Chand]

---
Message: 10
Date: Wed, 5 May 2010 10:26:34 -0500
From: Tilghman Lesher tles...@digium.com
Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 201005051026.34929.tles...@digium.com
Content-Type: text/plain;  charset=iso-8859-1

On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote:
 I can connect to the database and run via isql, and also use
func_odbc,
 etc with res_odbc configured with the same database / freetds, but I
 cannot write CDRs.

Are you writing to the database with func_odbc, or just reading?  My
gut says
that you need to check your permissions on the database to ensure that
you're
allowed to write to the CDR table.


Hi Tilghman, yeah I thought so too at first but then, using the
same permissions I'm doing both read  writes as well. 

On the database end, the user is setup as database_owner and has db_read
 db_write permissions.

I got Leif to check this with me last night, we couldn't figure it out. 

The error that pops up is: 

cdr_odbc: Connected to asterisk-freetds-connector
cdr_odbc: Error in PREPARE -1
cdr_odbc: Query FAILED Call not logged!



__


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Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install

[Neeraj Chand]

Use kickstart to configure your default packages, and then set up a
shell script to install the additional stuff you need. 

:)

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[asterisk-users] Asterisk realtime chat

[Neeraj Chand]

Hi,

Do a google search for openfire + asterisk.

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[asterisk-users] MYSQL queries from dial plan

[Neeraj Chand]

Hi all, 

I currently run small scale mysql queries from the dialplan 

exten = s,n,MYSQL(Connect

exten = s,n,MYSQL(Query resultid ${connid}

exten = s,n,MYSQL(FETCH fetchid

exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})


This currently takes about 4 seconds to complete.

If I run two simultaneous queries, this goes up to about 9 seconds for
both queries to complete. 

Is there a way that I can bring this time down? 

What sort of time delays are there if I use func_odbc? 

Thanks, 

Neeraj

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Re: [asterisk-users] FAX for Asterisk

[Neeraj Chand]

Just finished with the instructions from digium website/ net on how to
compile FFA:

After restart, modules did not get loaded so tried to load manually: 

[Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module:
Error loadin ile: No such file or directory
[Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module
'res_fax.so

Verified the files exist:

astbh00*CLI module load res_f
res_fax.so res_features.so res_fax_digium.so
astbh00*CLI module load res_f


Help! 

:)

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Re: [asterisk-users] asterisk-users Digest, Vol 65, Issue 38

[Neeraj Chand]

Did you check the jitter settings on asterisk  the phones as well? 

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Re: [asterisk-users] Extension in use

[Neeraj Chand]

There are a couple of ways you could see that, 

One would be by having a service .NET connected to the manager interface
and watching for activity on the phone, this way you could tell if the
phone is busy or not. 

[If phone has more than one line then set call-limit=1]

Is this for routing purposes or just for display? 

The other thing you could use is Jabber. 

Look for OpenFire integration with asterisk and you'll see what I mean
[google]

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Re: [asterisk-users] CDR userfield -

[Neeraj Chand]

Set(CDR(userfield|r)=blah)

This works for me on 1.4.24

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Re: [asterisk-users] odbc to ms-sql server

[Neeraj Chand]

Hi all, 

I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install

My problem is that I cannot get asterisk to build func_odbc 
res_odbc.so

I installed yum -y install unixODBC unixODBC-devel libtool-ltdl
libtool-ltdl-devel

And then went on to reconfigure / recompile asterisk

after a ./configure --with-odbc=/usr/lib/

I get 
###
checking for mandatory modules:  UNIXODBC... ok
configure: creating ./config.status


And then when I go to make menuselect;

[XXX]Res_odbc 

[XXX] func_odbc

[XXX] cdr_odbc

Can anyone help out with what I am missing? 

[I've gotten to a stage where tsql and isql connections to my sql db
work, however, getting odbc right is making me pull my hair out a bit]

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Re: [asterisk-users] odbc to ms-sql server

[Neeraj Chand]

 Gotcha! Missed libtool! :)

-Original Message-
From: Neeraj Chand 
Sent: Friday, 6 November 2009 6:43 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: odbc to ms-sql server

Hi all, 

I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install

My problem is that I cannot get asterisk to build func_odbc 
res_odbc.so

I installed yum -y install unixODBC unixODBC-devel libtool-ltdl
libtool-ltdl-devel

And then went on to reconfigure / recompile asterisk

after a ./configure --with-odbc=/usr/lib/

I get
###
checking for mandatory modules:  UNIXODBC... ok
configure: creating ./config.status


And then when I go to make menuselect;

[XXX]Res_odbc 

[XXX] func_odbc

[XXX] cdr_odbc

Can anyone help out with what I am missing? 

[I've gotten to a stage where tsql and isql connections to my sql db
work, however, getting odbc right is making me pull my hair out a bit]

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Re: [asterisk-users] Astreicon presentations

[Neeraj Chand]

Hi Folks, 

Are all the astricon presentations up? 

I'm especially after the one that tilghman did. I caught the tail end of
the prez when I decided to skip the session I was attending and go for
that one. 

:)

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Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in

[Neeraj Chand]

Please post your dial peer configurations. 

We have as5400 (5) working with asterisk servers also. 

The cisco routers are at the edge of the network (connected to PSTN via
E1) and send calls to asterisk over SIP 

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Re: [asterisk-users] SIP debugging enabled : written to log

[Neeraj Chand]

There was a presentation at astricon by Clod, that covers just this CLI 
Filters 

What this does is show only the filters that you set on asterisk cli, and your 
/var/log/asterisk/full log file also only contains the filtered output. 

I believe it would have been handier to have filtering, but with everything 
going to the full log so that if we need to debug in greater detail / look at 
events outside the scope of thep filters we set, it would be available in the 
full logs.

 

But thats my personal perspective. This may be useful to you. The presentation 
should be on astricon.net some time soon

 

Cheers!

 

 

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Re: [asterisk-users] DUNDi + SIP Realtime

[Neeraj Chand]

JR - couldn't find your whitepaper from astricon06 online, links are
broken would it be possible for you to email it to me? 

I have not tried setting up DUNDi yet, but from the sound of it, seems
like it would be pretty handy. 

I have sip phones registering to two asterisk servers [primary and
backup] and have a macro setup where incoming call checks if sip phone
is available via ${SIPPEER}(${EXTEN}:status and if it is not on the
primary server, then sends the call to the backup server in a separate
context which attempts to dial the sip phone once, and if the phone is
offline / unavailable there as well, then to send it to voicemail or
followme as the case may be. 

However, DUNDi sounds better in terms of scalability as I'm fast
outgrowing two servers. 

:)

Thanks, 

Neeraj

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Re: [asterisk-users] Custom auto-install asterisk using ks.cfg

[Neeraj Chand]

Hi guys, 

Anyone done this with CentOS and asterisk 1.4? 

thanks


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Re: [asterisk-users] dCAP Exam

[Neeraj Chand]

Hi All,

Thanks for all the wonderful contributions, from cell phones right up to
proxies, etc...

Many thanks also to Tony Turner for the great advice.

As for Jared, what can I say...simply legend... :) 

I believe this is what I was after.

:)

For all those attending AstriconSee you there! 




 The most helpful thing would be a past scenario, something that has
come
 up in previous dCAP exams.
 
 Can anyone send in a short descriptor of the final prac [real scenario
 that has happened before?]



Without going into too much detail on the exact details of the dCAP
exam, the general idea is this:  A small company has hired you to build
a typical small-business PBX using Asterisk, and you have 90 minutes to
get it up and running.  Given the time constraint, we really stick to
the basics, so there shouldn't be anything unexpected during the test.


-- 
Jared Smith
Training Manager
Digium, Inc.




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Re: [asterisk-users] dCAP Exam

[Neeraj Chand]

Hmm...so by open book, that means access to the internet? Possible to
get own notes ? 

The most helpful thing would be a past scenario, something that has come
up in previous dCAP exams.

Can anyone send in a short descriptor of the final prac [real scenario
that has happened before?]

I'm not worried about the content, it's an example that I'm after, so
that I don't get thrown by it. :)

Thanks Tilghman, :)

_
Date: Tue, 15 Sep 2009 08:38:04 -0500
From: Tilghman Lesher tles...@digium.com
Subject: Re: [asterisk-users] dCAP Exam
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 200909150838.05001.tles...@digium.com
Content-Type: text/plain;  charset=iso-8859-1

On Monday 14 September 2009 22:31:26 Neeraj Chand wrote:
 Is there anywhere I can possibly get a model of the exam itself, maybe
 possible scenarios for the prac, etc?

The practical is open-book.  You're welcome to look up anything you
want, but
the time constraint pretty much guarantees that if you know what you're
doing,
you can pass the practical.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] dCAP Exam

[Neeraj Chand]

Hi folks, 

Is there anywhere I can possibly get a model of the exam itself, maybe
possible scenarios for the prac, etc? 

To people who have done the examany helpful hints ? 

Thanks,

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Re: [asterisk-users] Time of Day Routing

[Neeraj Chand]

Asterisk version 1.4



From: Neeraj Chand 
Sent: Friday, 14 August 2009 8:17 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Time of Day Routing

 

Hi David,

 

With this: 

   ifTime(00:00-12:00|*|*|*)

 

Whatever time you specify at the end, I believe asterisk continues to
evaluate this condition as true for 2 more minutes.

 

So in this case, it will be valid for 00:00-12:02, even though you've
specified 12:00

 

Cheers!

 

Neeraj

 

 

 

  

 

 

 

 

 

 

 

 

 

 

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[asterisk-users] Time of Day Routing

[Neeraj Chand]

Hi David,

 

With this: 

   ifTime(00:00-12:00|*|*|*)

 

Whatever time you specify at the end, I believe asterisk continues to
evaluate this condition as true for 2 more minutes.

 

So in this case, it will be valid for 00:00-12:02, even though you've
specified 12:00

 

Cheers!

 

Neeraj

 

 

 

  

 

 

 

 

 

 

 

 

 

 

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[asterisk-users] Astricon 2009 - dCAP

[Neeraj Chand]

Hi folks, 

 

Going to astricon this year? Feeling a bit nervous as planning to take
the exam this time. Any one else doing the same? 

 

 

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[asterisk-users] FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan

[Neeraj Chand]

Managed to implement this on asterisk v1.4.24.1, 

Also, Hangupcause updating to user field. 

However, this only works on the edge of my voice network (demarcation
point) 

It does not work on my internal routing boxes as I use IAX to route
between remote sites.

I was thinking of using some sort of SIP variables to transport these
results over the IAX trunk.. 

Any bright ideas folks???

[outbound]

exten = _XXX.,n(dial),gotoif($[${loadbalance} = 1 ]?balance) ;at dial
decide if we want to load balance

exten = _XXX.,n,goto(50)   ;if no balance then outbound call
initiated with failover

exten = _XXX.,n(balance),NoOp(Load Balancing Active)
;NoOp - display Load balancing status

exten = _XXX.,n,Set(counter1=${DB(data/counter)})
;retrieve counter value from astdb

exten = _XXX.,n,NoOp(${counter1})
;display value of counter

exten = _XXX.,n,Set(counter=${MATH(${counter1}+1,int)})   ;
increment counter

exten = _XXX.,n,Set(DB(data/counter)=${counter})
;write incremented value back to asterisk db

exten = _XXX.,n,Set(result=${MATH(${counter}%2)})
;check for Odd/ Even using modulus of ${counter} via MATH function

exten = _XXX.,n,NoOp(${result})
;Display output - 0 for even and 1 for odd

exten = _XXX.,n,GotoIf($[${result}  0]?50:100);Odd calls route to
outbound-1, even calls to outbound-2

exten = _XXX.,n(balance),NoOp(Load Balancing Active) ;NoOp -
display Load balancing status

exten = _XXX.,n,Set(counter1=${DB(data/counter)})
;retrieve counter value from astdb

exten = _XXX.,n,NoOp(${counter1})  ;display
value of counter

exten = _XXX.,n,Set(counter=${MATH(${counter1}+1,int)});
increment counter

exten = _XXX.,n,Set(DB(data/counter)=${counter})   ;write
incremented value back to asterisk db

exten = _XXX.,n,Set(result=${MATH(${counter}%2)})  ;check
for Odd/ Even using modulus of ${counter} via MATH function

exten = _XXX.,n,NoOp(${result})
;Display output - 0 for even and 1 for odd

exten = _XXX.,n,GotoIf($[${result}  0]?50:100);Odd calls route
to outbound-1, even calls to outbound-2

 

exten = _XXX.,50,gotoif($[${dialout}  0 ]?firstfail)

exten = _XXX.,n,Set(dialout=${MATH(${dialout}+2)})

exten = _XXX.,n,NoOp(${dialout})

exten = _XXX.,n,dial(${route2}/${ext...@${context2})

exten = _XXX.,n,goto(after-dial)

exten = _XXX.,n(firstfail),set(try=${MATH(${dialout}+2)})

exten = _XXX.,n,dial(${route2}/${ext...@${context2})
;attempt to dial out via route 2

exten = _XXX.,n,goto(after-dial);after attempting to dial, go to
after-dial

 

 

exten = _XXX.,100,gotoif($[${dialout}  0 ]?secondfail)

exten = _XXX.,n,Set(dialout=${MATH(${dialout}+1))

exten = _XXX.,n,NoOp(${dialout})

exten = _XXX.,n,dial(${route1}/${ext...@${context1})
;attempt dial out via route 1

exten = _XXX.,n,goto(after-dial)
;after attempting to dial, go to after-dial

exten = _XXX.,n(secondfail),set(try=${MATH(${dialout}+2})
;If call has been dialled by other route and is failing over, set
variable try = 2

exten = _XXX.,n,dial(${route1}/${ext...@${context1})
;attempt to dial out via route 1

exten = _XXX.,n,goto(after-dial)
;after attempting to dial, go to after-dial

 

exten = _XXX.,n(after-dial),Set(CDR(accountcode)=${DIALSTATUS})
;first step - add ${DIALSTATUS} to CDR in accountcode field

exten = _XXX.,n,goto(${DIALSTATUS})
;go to dial-status received from attempt

 

exten = _XXX.,n(BUSY),,goto(set_cause)
;if Busy, go to set_cause

exten = _XXX.,n(NOANSWER),goto(set_cause)
;if No answer, go to set_cause

exten = _XXX.,n(CANCEL),goto(set_cause)
;if Cancel, go to set_cause

exten = _XXX.,n(NOANSWER),goto(set_cause)
;if No Answer, go to set_cause

 

 

exten = _XXX.,n(CHANUNAVAIL),gotoif($[${try} = 2 ]?emergency-notify)
;If CHANUNAVAIL,check if both routes are down. If yes, send emergency
notification

exten = _XXX.,n,gotoif($[${first-dial} = ${route1} ]?notify-1)
;If this was first attempt from route 1, go to notification for route 1
down

exten = _XXX.,n,goto(notify-2)
;else go to notification for route 2

 

exten = _XXX.,n(set_cause),hangup()
;at set_cause, firstly hang up channel (if not done already)

exten = _XXX.,n,goto(CDRfield)
;go to CDR field mapping section

 

exten = _XXX.,n(CDRfield),Set(CDR(userfield)=${HANGUPCAUSE})
;set Hangupcause to user field in CDR

 

exten = _XXX.,n(notify-1),System(echo Call redirect detected on
'${route1}'  | mail -s Calls Fail Over neeraj.ch...@ocis.com.au)
;send notification route 1 down

exten = _XXX.,n,goto(100)  ;attempt dial via route 2

 

exten = _XXX.,n(notify-2),System(echo Call redirect detected on
'${route2}'  | mail -s Calls Fail Over neeraj.ch...@ocis.com.au)
;send notification route 2 down

exten = _XXX.,n,goto(50)  ;attempt dial via route 1

 

exten = _XXX.,n(emergency-notify),System(echo Call redirect detected
on BOTH Routes!  | mail -s Calls Fail Over
neeraj.ch...@ocis.com.au) ;send critical - both routes

exten = _XXX.,n,goto(set_cause)   

Re: [asterisk-users] Writing Hangup causes to CDR record

[Neeraj Chand]

Hi guys, 

I'm trying to write hangup causes from asterisk into the CDR record.

Using version 1.4.24.1 at the moment, but no joy so far.

Has anyone implemented this? 


Neeraj Chand
Support Analyst 

Fiji Islands Australia  
T: +6793342526   T: +61388924326
M:+6799344012New Zealand
www.ocis.com.au  T: +649 980 7022   

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-users-requ...@lists.digium.com
Sent: Thursday, 21 May 2009 8:28 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 58, Issue 56

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

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Today's Topics:

   1. Re: DAHDI fun and games (Danny Nicholas)
   2. Re: Step-by-Step Asterisk and MeetMe Help (Tzafrir Cohen)
   3. Re: Channels configuration with DAHDI (Dave Fullerton)
   4. Re: ...is circuit busy message (Jeff LaCoursiere)
   5. Re: Dialplan Priorities and Sort Order... (Alex Samad)
   6. Re: Step-by-Step Asterisk and MeetMe Help (Jimmy Ezell)
   7. Re: Open source SIP client (marek cervenka)
   8. Re: Step-by-Step Asterisk and MeetMe Help (Jonathan Thurman)
   9. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business)
  10. Re: Channels configuration with DAHDI (Daniel Bareiro)
  11. 1.4.24.1 - 1.6.0.9: segfault (sean darcy)
  12. Voicemail playback NEWEST first vs. OLDEST first (Karl Fife)
  13. Re: Step-by-Step Asterisk and MeetMe Help (Jeff LaCoursiere)
  14. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business)
  15. Bridging INBOUND PRI to OUTBOUND PRI fails with   Monitor()
  (Barry L. Kline)
  16. PSTN Connection (Manoj Panicker - FOES)
  17. Re: Open source SIP client (Alex Samad)
  18. Re: PSTN Connection (Paul Hales)
  19. interruption in queue (Rilawich Ango)
  20. Re: PSTN Connection (--[ UxBoD ]--)
  21. Polycom Productivity Suite (Matt Darnell)
  22. Fwd: Asterisk CCM, CME Integration (Arun Kumar)


--

Message: 1
Date: Wed, 20 May 2009 16:07:48 -0500
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] DAHDI fun and games
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 2897b95e2e394a7d9fad95bff31bf...@db0002
Content-Type: text/plain;   charset=us-ascii

Using r/m because DAHDI takes 10-15 seconds to get TELCO rings.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave
Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI fun and games

Danny Nicholas wrote:
 Hi Listers,
 
I'm running 1.4.25-rc1 on opensuse 11.0 with
 dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and
snapdsp.0.0.2.
 Incoming calls work fine.  Outgoing calls made directly (exten =
 s,1,Dial(DAHDI/G1) then number work fine.  The problem I have is
trying to
 let Asterisk make the call (exten = s,1,Dial(DAHDI/G1/5551212,,r).
If I
 use m (moh) the music plays 5-8 seconds after the other end picks
up.
 When using r, I get 2-3 rings after other end picks up.  I've went
through
 every flavor of dahdi-linux from 2.0.0 to 2.1.0-rc4 (which crashed me)
with
 no joy.  Any suggestions?   Hardware is Dell Poweredge 1650/1550 and
 TDM410P/TDM400P.

Any reason you're using the r/m option at all? Since this is an analog 
card I would leave the r/m off and just let asterisk use the in-band 
progress from the telco.

-Dave

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Message: 2
Date: Thu, 21 May 2009 00:11:24 +0300
From: Tzafrir Cohen tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: asterisk-users@lists.digium.com
Message-ID: 20090520211124.gm3...@xorcom.com
Content-Type: text/plain; charset=us-ascii

On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:

 multi-processor machine  ( I had to remember to specify smp for the
kernel)

I repeat: why bother with such an old system? Really?

Recall the comment from the book. That book had nothing really specific
to Centos 4. Why do you

[asterisk-users] FW: Writing Hangup causes to CDR record

[Neeraj Chand]

Hi guys, 

I'm trying to write hangup causes from asterisk into the CDR record.

Using version 1.4.24.1 at the moment, but no joy so far.

Has anyone implemented this?

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[asterisk-users] Fwd: Asterisk With Cisco Voice Router

[Neeraj Chand]

Hi, 

We have AS5400's set up with asterisk boxes. Initially we had similar
issues, but as described, you need to have dial peers to handle both
incoming and outgoing peers.

Please post your dial peer configs as well as the serial interface
configs. I also found that until I add [isdn incoming-voice modem ] I
could not get incoming calls on that serial interface to route to my *
box.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-users-requ...@lists.digium.com
Sent: Saturday, 16 May 2009 10:00 AM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 58, Issue 40

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
than Re: Contents of asterisk-users digest...


Today's Topics:

   1. Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
   2. Re: Fwd: Asterisk With Cisco Voice Router (Steve Howes)
   3. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
   4. Re: Fwd: Asterisk With Cisco Voice Router (David Backeberg)
   5. Re: meetme dies looking for conf-getconfno (sean darcy)
   6. howto set up persistent dynamic meetme (sean darcy)
   7. Agent-Login/out in 1.6 (David Anthony O Reilly)
   8. Agent-Login/out in 1.6 (David Anthony O Reilly)
   9. Re: Agent-Login/out in 1.6 (Stefan Reuter)
  10. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
  11. Re: Agent-Login/out in 1.6 (Jim Dickenson)
  12. Re: howto set up persistent dynamic meetme (Tilghman Lesher)
  13. Re: Fwd: Asterisk With Cisco Voice Router (Philipp Kempgen)


--

Message: 1
Date: Sat, 16 May 2009 14:46:27 +0300
From: Timothy Smith timotsm...@gmail.com
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
416fc8170905160446r5815fd87m67e62506ad9ac...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Hi,

In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware ?as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.

I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!

Has anyone had the above set up working successfully? Attached are some
confs.

Thanks a lot for your assistance.

Kind Regards,
Wilson
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-- next part --
cs-intranet*CLI sip show peers
Name/username  HostDyn Nat ACL Port Status
103172.17.3.2495060 OK (3
ms)
102172.17.3.2485060 OK (3
ms)
101172.17.10.150   5060 OK (1
ms)
100/100172.19.4.102 D   N  32544
Unmonitored
4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0
offline]


; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP
phone
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--

Message: 2
Date: Sat, 16 May 2009 13:25:40 +0100
From: Steve Howes st...@geekinter.net
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 659dc612-4035-4d7e-a73c-77b5a16d6...@geekinter.net
Content-Type: text/plain; 

Re: [asterisk-users] asterisk-users Digest, Vol 58, Issue 9

[Neeraj Chand]

--- SIP read from 192.168.32.245:5060 ---
SIP/2.0 481 CallLeg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport
From: asterisksip:aster...@192.168.32.16;tag=as2ff08179
To: sip:5...@192.168.32.245:5060;user=phone;tag=c0a80101-2ce1bc03
Call-ID: 2fa28b4-c0a80101-d-9...@192.168.32.245
CSeq: 143 NOTIFY
Content-Length: 0





Reliably Transmitting (no NAT) to 192.168.32.245:5060:
NOTIFY sip:5...@192.168.32.245:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport
From: asterisk sip:aster...@192.168.32.16;tag=as2ff08179
To: sip:5...@192.168.32.245:5060;user=phone;tag=c0a80101-2ce1bc03
Contact: sip:aster...@192.168.32.16
Call-ID: 2fa28b4-c0a80101-d-9...@192.168.32.245
CSeq: 143 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:aster...@192.168.32.16
Voice-Message: 0/2 (0/0)

Can anyone help me out with this? 

I just recently upgraded to asterisk 1.4.24.1. 

Use Thomson ST2030s sip phones. 

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