Re: [asterisk-users] Confbridge

[Sam Basan]

John,

What you see it's how it should be.

The wait for admin means that all users join the conference room but the
conference is not started and they all should hear MOH.

When the admin will join then the conference will start and all will hear
the admin (or all others if they are not muted)

 

 

 

 

 

 



 

Office:  +972-77-6662000  E-mail:
 off...@bluebe.net 

Fax:   +972-77-6662020  Web:
 http://www.bluebe.net   

 



 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of John T. Bittner
Sent: Saturday, August 8, 2020 12:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion‏

Subject: [asterisk-users] Confbridge

 

To all:


No matter what I try, I cannot get the system to wait for the admin to join.
It just dumps users into the bridge directly.

I do not have a pin for users, does that matter?

 

What am I missing?

 

Another issue the absolute timeout is not working ? … have recordings that
last for over 24 hours… and this should not happen…

All calls should hangup after 4 ?

 

Any ideas ?

 

Any help is much appreciated.

 

Thanks

 

This is my dialplan.

 

exten => s,1,Wait(1)

exten => s,n,Answer

exten => s,n,Set(TIMEOUT(absolute)=14400)

exten => s,n,NoOp(${CALLERID(name)})

exten => s,n,NoOp(${CALLERID(num)})

exten => s,n,NoOp()

exten => s,n,Playback(church) ; "Please hold while..."

exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no)

exten => s,n,Set(CONFBRIDGE(user,startmuted)=yes)

exten => s,n,Set(CONFBRIDGE(user,template)=church)

exten => s,n,Set(CONFBRIDGE(user,marked)=no)

exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes)

exten => s,n,Set(CONFBRIDGE(user,end_marked)=yes)

exten => s,n,ConfBridge(xaccel)

exten => s,n,hangup

 

confbridge.conf

 

[general]

[church]

type=user

startmuted=yes

announce_join_leave=no

announce_user_count=no

wait_marked=yes

end_marked=yes

music_on_hold_when_empty=no

quiet=yes

;

[xaccel]

type=bridge

record_conference=yes

;

Then calling in I see this 

Conference Bridge Name   Users  Marked Locked Muted

 == == == =

xaccel1  0 No No

 

 

John Bittner

CTO



380 US Highway 46, Suite 500

Totowa, NJ 07512

Phone: 201.806.2602 x2405

Fax:   201.806.2604

Cell:   973.390.1090

  www.xaccel.net

 

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the
sender by reply e-mail and destroy all copies of the e-mail.

 

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Re: [asterisk-users] SIP Codec negotiation

[Sam Basan]

The unwritten rule of SDP is that if possible you use the first codec of a
type listed, but you don’t have to.  If the sender says he can do
something, he had better be prepared to handle media of that type no matter
in what order it was listed.

So when you send OK with ulaw as first priority and get ACK most probably
it will be ulaw

בתאריך יום ו׳, 11 במאי 2018, 16:08, מאת Steve Edwards ‏<
asterisk@sedwards.com>:

> > On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote:
>
> >> I receive an INVITE/SDP containing:
> >>
> >>  m=audio 11310 RTP/AVP 3 0 101
> >>
> >> which I interpret as gsm, ulaw, rfc2833.
> >>
> >> and I reply with an OK/SDP containing:
> >>
> >>  m=audio 15884 RTP/AVP 0 3 101
> >>
> >> which I interpret as ulaw, gsm, rfc2833.
> >>
> >> How can I tell which codec was actually used for the call?
>
> On Fri, 11 May 2018, Daniel Tryba wrote:
>
> > AFAIK this is undetermined. The callee can send either ulaw or gsm,
> > unless the caller wants to narrow it down to 1 codec, see
> > https://tools.ietf.org/html/rfc4317#section-2.2
> >
> > Most of the time the callee will pick the first (so in this case ulaw).
> > But there are media gateways out there that choose g711[au] above "more
> > complex" codecs regardless order in SDP. My prefer PSTN provider will
> > always prefer alaw if offered since that will prevent transcoding on
> > their side if the call goes to ISDN/POTS, but AMR if the call goes to
> > VoLTE.
>
> So, without examining the RTP, you cannot tell which codec was actually
> used?
>
> In the above example, even though the INVITE/SDP says they prefer gsm over
> ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose to use
> gsm or ulaw?
>
> Can it be asymmetrical? They send gsm and I send ulaw?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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Re: [asterisk-users] how to make International calls from asterisk PBX

[Sam Basan]

International calls are exactly as local phones using the same lines/trunks.

First check your outbound route to verify that your dial plan match your 
dialing international pattern.







Sincerely,



Sam Basan





From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Uzma Anjum
Sent: Monday, February 12, 2018 1:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to make International calls from asterisk PBX



Hello...



I'm running asterisk-13 and international calls are not working in it.How can I 
make it work.Can anyone please tell me.



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https://www.avast.com/antivirus
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Re: [asterisk-users] asterisk-users Digest, Vol 160, Issue 5

[sam habash]

Get Outlook for Android

From: asterisk-users-boun...@lists.digium.com 
 on behalf of 
asterisk-users-requ...@lists.digium.com 

Sent: Monday, November 6, 2017 6:00:01 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 160, Issue 5

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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You can reach the person managing the list at
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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. ? PJSIP and Non Media Proxy (Saint Michael)
   2. Re: ? PJSIP and Non Media Proxy (Joshua Colp)
   3. Re: ? PJSIP and Non Media Proxy (Pete Mundy)
   4. PJSIP console messages with Zoiper (Brian Capouch)
   5. Re: PJSIP console messages with Zoiper (Richard Mudgett)


--

Message: 1
Date: Sun, 5 Nov 2017 13:42:50 -0500
From: Saint Michael 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] ? PJSIP and Non Media Proxy
Message-ID:

Content-Type: text/plain; charset="utf-8"

?Now that Joshua had the kindness to respond, I see here a big disconnect
between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like me
would need to avoid proxying the media.  Would would Digium support and
bring in with such fanfare a channel like PJSIP that lacks the only thing
that 99% would need to do business in an efficient manner? I mean people
like me buy and sale billion of minutes every day, and most of my peers
gravitate towards Opensips and other solution that do not touch the media.
Yesterday I had to roll back my sleeves and go back to the old sip channel.
I would love to see Asterisk-PJSIP to find a way to act like a proxy. This
would turn Asterisk into a real wholesale business tool, which is not, so
far.
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Message: 2
Date: Sun, 05 Nov 2017 15:16:09 -0400
From: Joshua Colp 
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ? PJSIP and Non Media Proxy
Message-ID:
<1509909369.1519452.1162348936.4eb33...@webmail.messagingengine.com>
Content-Type: text/plain; charset="utf-8"

On Sun, Nov 5, 2017, at 02:42 PM, Saint Michael wrote:
> ?Now that Joshua had the kindness to respond, I see here a big disconnect
> between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like
> me
> would need to avoid proxying the media.  Would would Digium support and
> bring in with such fanfare a channel like PJSIP that lacks the only thing
> that 99% would need to do business in an efficient manner? I mean people
> like me buy and sale billion of minutes every day, and most of my peers
> gravitate towards Opensips and other solution that do not touch the
> media.
> Yesterday I had to roll back my sleeves and go back to the old sip
> channel.
> I would love to see Asterisk-PJSIP to find a way to act like a proxy.
> This
> would turn Asterisk into a real wholesale business tool, which is not, so
> far.

It's not the lack of this feature which drives people to using OpenSIPS
or Kamailio for this use case. It's just fundamentally designed
differently and better performant for that scenario. Asterisk isn't the
best solution for everything everyone needs or wants, and that's okay.
There are other projects (like those already mentioned) that are a
better fit, and Asterisk can even play a part in there as an application
server.

I'm a firm believer in using the right tool for the right job even if it
means that Asterisk isn't the right fit. Frustrated users are something
I never want to see.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & 
www.asterisk.org



--

Message: 3
Date: Mon, 6 Nov 2017 08:37:30 +1300
From: Pete Mundy 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] ? PJSIP and Non Media Proxy
Message-ID: <958b7a39-4b14-4c08-a508-6c7dd3cd5...@fiberphone.co.nz>
Content-Type: text/plain; charset="us-ascii"

> On 6/11/2017, at 7:42 AM, Saint Michael 

[asterisk-users] Tranfer the called number in 3 way call

[sam habash]

Hey there,


I have a question i want a dialplan to send the called number of the client 
instead of my callerID when making a 3way call or when transfering to an 
extension from a bridge to another pbx. The problem i add a variable and using 
thw two underscores but i still see the my calledID , i am using both asterisk 
1.8.29 and other is asterisk 11.4 here is the dialplan inherit which i do :


exten=> _6XX,n,set(__var=${EXTEN})

exten=> _6XX,n,Dial(IAX2/bridge/${EXTEN},,tTor)


I want to recieve the client number that were called from my first pvx with 5XX 
extensions to be shown on my second pbx with 6XX while making a transfer to the 
bridge with 3 way call or blind transfer , I know i am missing something here , 
can you guys help me.

Sent from my LG Mobile
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Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

[Sam Basan]

Your second call is not without sound, there is simply no call at all.
As the first answer the call his channel and the external call channel
connected.
The second device simply off hook but his channel have no external channel
to connect.

It's looks like a simple telephony glare.

Sam

בתאריך 30 בנוב' 2016 7:00 PM,‏ "marek cervenka" <cerva...@gmail.com> כתב:

> hi,
>
> our customer reports problem when 2 agents answer the call in the same time
>
> faster operator (device) answer the call, but the second is showed up (on
> device) and call is without sound
>
> asterisk 13.9/app_queue with strategy ringall/operators via Local channel
> with sip device (chan_sip)
>
> do you have any tips/info before i will dig deep into logs/debug?
>
> checked google without any clue
>
> marek
>
>
>
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Re: [asterisk-users] How to custom the message on call busy or no answer in asterisk

[Sam Basan]

Plays a tone list, either the tone named tonename defined in the  
<http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf> 
indications.conf file, or a directly specified tonelist of frequencies and 
durations. See 
<http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf> 
indications.conf for a description of the specification of a tonelist.

 

The second line play the busy tone.

The third line send a busy response back to the request message.

 

You can also play a sound file with busy or other requested sound instead of 
using a tone

 

 

 

Sincerely,

 

Sam Basan



 

From: Thyda ENG [mailto:ength...@gmail.com] 
Sent: Saturday, November 21, 2015 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] How to custom the message on call busy or no 
answer in asterisk

 

I found this but I don't know where the busy tone place, I wanna replace this 
file, do you have any idea ?

​

 
<https://drive.google.com/file/d/0B2n_BStebemaWmtNR1JNSXY4STg/view?usp=drive_web>
  Screen Shot 2015-11-21 at 1.49.47 PM.png

​

 

On Fri, Nov 20, 2015 at 1:51 PM, Julien Sansonnens <jul...@jsansonnens.ch 
<mailto:jul...@jsansonnens.ch> > wrote:

Hi,
Check the DIALSTATUS variable.
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS

Regards, Julien

--




2015-11-20 2:15 GMT+01:00 Thyda ENG <ength...@gmail.com 
<mailto:ength...@gmail.com> >:
> Hi,
>
> I was wonder is there any way to custom the message on the call busy or no
> answer I actually get the error code from asterisk server on busy or no
> answer. Can I custom the text message or custom the message to sound ?
> Anyone have any idea could u please share me ?
>
>
> Thank,
>
> Thyda
>

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Re: [asterisk-users] No sound with internal calls depending on which phones

[Sam Basan]

Snom default configuration is SRTP enabled.

You should disable the SRTP from the phone web GUI configuration

 

 

 

Sincerely,

 

Sam Basan



 

From: Mitul Limbani [mailto:mi...@enterux.in] 
Sent: Thursday, November 12, 2015 5:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] No sound with internal calls depending on which 
phones

 

You might have to disable srtp negotiations inside the phone web ui options. 

Mitul 

On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbuche...@hsolutions.ch 
<mailto:dbuche...@hsolutions.ch> > wrote:

Dear all,

I have a very strange problem :

*   external calls work perfectly,
*   internal calls between some phones too,
*   but internal call between two similar phones don't work !!! (Snom 710)

When we have sound, there are no errors in asterisk. When we do not have sound, 
there is the following error :

*   [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP 
module loaded, can't setup SRTP session.

This is a working internal call :



  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1") in 
new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Got  RTP packet from192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 126, seq 031575, ts 01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown 
RTP codec 126 received from '192.168.128.99:49646 <http://192.168.128.99:49646> 
'
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on 'SIP/dbucher-'

This is a non-working call :



  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module 
loaded, can't setup SRTP session.
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", "SIP/phone1") 
in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494 <http://192.168.128.228:65494>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 <http://192.168.128.228:65494>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/hsolutionspf5-0002'

I tried many options to disable SRTP but without success :

*   canreinvite = no
*   canreinvite = nonat
*   srtpcapable=no
*   encryption=

[asterisk-users] Update new IP address (move temporarily) for INVITE

[Sam Basan]

Hello,

 

How can I update asterisk to send back move temporarily with updated IP
address to incoming INVITE.

i.e, Incoming call from ITSP to server 1 with x DID and there is a need to
update the ITSP that the specified x DID number is allocated in server 2.

 

Thanks,

Sam Basan

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[asterisk-users] Reverse one way paging or silent monitoring

[Sam Basan]

Hello,

 

Paging a phone set the phone to auto answer and open the speaker.

How can I set the phone to turn on just his microphone and the camera, if
available, for security remote controlling?

 

 

Thanks,

Sam

 

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Re: [asterisk-users] Help with voicemail

[Sam Basan]

Check your phone codecs.
It set to g729 while you don't have this codec in your asterisk nor files
in this codec.
בתאריך 17 באוק' 2015 18:34,‏ "Luca Bertoncello"  כתב:

> Hi list!
>
> My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
> voicemail.
> On two of these numbers the voicemail works without any problem, on the
> other
> it doesn't...
> I get this error:
>
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to
> open /var/spool/asterisk/voicemail/default/0039015111/unavail (format
> 0x100 (g729)): No such file or directory
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to
> open beep (format 0x100 (g729)): No such file or directory
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format:
> wav, 0x6edbd8
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format:
> gsm, 0x7c6978
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x40 (slin)
>
> Of course, I have a
> file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm...
>
> Can someone help me to solve my problem?
>
> Thanks a lot!
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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[asterisk-users] Asterisk AMI events filtering

[Sam Basan]

 

Hi folks,

 

I have one server with multiple companies (multi-tenant).

>From AMI I get all events of all extensions so any one that connect can see
other extensions, from different company (context).

How can I limit specific user to get just specific context?

 

Sam

 

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Re: [asterisk-users] SIP domain different than provider's

[Sam]

On 08/21/2015 12:52 AM, Sam wrote:

Hello,

I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains before. My user '1...@4354766787.com' is
handled by a 3rd party provider: 'sip.provider.com' and my local domain
on my asterisk box is the hostname 'mypbxdomain.com'.

My normal extension I use for everything is just '111'. I figured the
best way of joining my asterisk box was to just hard code in the
extensions I would need to dial for my remote office work (there are
only a couple of extensions so shouldn't be a big deal).

However I struggled to get authentication working for outgoing calls to
the few new extensions at the remote office through their provider.
Looking at debug logs it was clear that the sip 'To' address was wrong.
It had the provider: To: sip:1...@sip.provider.com instead of the
domain which should look like: To: sip:1...@4354766787.com (right?)

In the end, after hours of googling, reading the docs on sip.conf
several times revealed a little spoke of '!' dialplan option. Simply
changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to
'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.

But this seems really hackish. Is this the right/only way? Or is just
having a provider and mismatched domains not really the norm?

I have an an anonymized log here: http://tinyurl.com/ouy2ajr



Regards,
Sam




So since no one has responded, this either means one of two things:
Either I am an idiot and missed something obvious and therefore no one 
wants to deal with me. Or this is indeed not something typical of an 
asterisk config.


Can someone at least point me to which? :)
As I mentioned, everything is working, it just doesn't feel right.


Regards,
Sam

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[asterisk-users] SIP domain different than provider's

[Sam]

Hello,

I have what I would think would be a common situation: I run asterisk at 
home simply as a land line. I started a new job working remotely and 
they gave me a SIP account with user name, domain, and proxy. I've never 
had to deal with sip domains before. My user '1...@4354766787.com' is 
handled by a 3rd party provider: 'sip.provider.com' and my local domain 
on my asterisk box is the hostname 'mypbxdomain.com'.


My normal extension I use for everything is just '111'. I figured the 
best way of joining my asterisk box was to just hard code in the 
extensions I would need to dial for my remote office work (there are 
only a couple of extensions so shouldn't be a big deal).


However I struggled to get authentication working for outgoing calls to 
the few new extensions at the remote office through their provider. 
Looking at debug logs it was clear that the sip 'To' address was wrong. 
It had the provider: To: sip:1...@sip.provider.com instead of the 
domain which should look like: To: sip:1...@4354766787.com (right?)


In the end, after hours of googling, reading the docs on sip.conf 
several times revealed a little spoke of '!' dialplan option. Simply 
changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to 
'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.


But this seems really hackish. Is this the right/only way? Or is just 
having a provider and mismatched domains not really the norm?


I have an an anonymized log here: http://tinyurl.com/ouy2ajr



Regards,
Sam

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Re: [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

[Sam Basan]

Hi,
You must have two thing for start:
1. Set your FW to allow sip port (by default 5060) to your asterisk IP
address.
2. Set your asterisk configuration with the external public IP and your
local subnet address (so asterisk will put his public address for outside
the networks calls)

Google for asterisk NAT configuration parameters.

נשלח מטלפון נייד
בתאריך 14 באוג' 2015 22:12,‏ Daniel - Asterisk earohua...@gmail.com כתב:

 Hello Sam,

 Do you have any recommendation to overcome these NAT issues?

 On 8/14/15, Sam Basan sba...@bluebe.net wrote:
  Hi,
 
  It's looks like you are having NAT problem.
  Packets from the provider fail reaching your box.
 
  נשלח מטלפון נייד
  בתאריך 14 באוג' 2015 15:56,‏ Daniel - Asterisk earohua...@gmail.com
  כתב:
 
  Hello friends:
 
  I am facing cutoffs randomly when negotiating calls.
 
  The PBX dials the destination, the provider (softswitch) receives the
  request *[1]* and sudenly the PBX hangs up the call* [2]* while the
  provider is still dialing it, as a consequence the remote peer receives
 a
  ghost call. Along the atempt I could see six times a messages regarding
  NAT
  isuues *[3]*
 
  I hope anyone can give me an idea to solve this issue. Softswitch is
  using
  an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with
  Asterisk 1.8.11.0
 
  Thanks in advance
 
  Elder D. Arohuanca
  Lima - Peru
 
 
  *[1]*
  [Aug 12 19:21:05] VERBOSE[17115] app_dial.c:-- Called
  SIP/SIP-PROVIDER/965034648
 
 
  *[2]*
  [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout
  reached
  on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for
 seqno
  103 (Critical Request) -- See
  https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  Packet timed out after 8832ms with no response
  [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call
  0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our
 critical
  packet (see
  https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  ).
  [Aug 12 19:21:14] VERBOSE[17115] app_dial.c:   == Everyone is
  busy/congested at this time (1:0/0/1)
  [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
  [s@macro-dialout-trunk:20] NoOp(SIP/143-01d8, Dial failed for
 some
  reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111) in new
 stack
  [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
  [s@macro-dialout-trunk:21] Goto(SIP/143-01d8, s-CHANUNAVAIL,1)
 in
  new stack
 
  *[3]*
  Retransmitting #3 (no NAT) to PROVIDER-IP:5060:
  INVITE sip:dialed_number@PROVIDER-IP SIP/2.0
  Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701
  Max-Forwards: 70
  From: PBX-DID sip:outbound-trunk@PROVIDER-IP;tag=as27ef83ae
  To: sip:dialed_number@PROVIDER-IP
  Contact: sip:outbound-trunk@PBX-PUBLIC_IP:5060
  Call-ID: 6b9ad82d4673fdab722f9e53411a767d@PROVIDER-IP
  CSeq: 103 INVITE
  User-Agent: FPBX-2.8.1(1.8.11.0)
  Proxy-Authorization: Digest username=outbound-trunk,
  realm=SoftSwitch,
  algorithm=MD5, uri=sip:dialed_number@PROVIDER-IP,
  nonce=d1b5806808a0888112190722408572932332,
  response=40c94f3c04e87e3382c7652d1f012dc9
  Date: Thu, 13 Aug 2015 00:56:40 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
 INFO,
  PUBLISH
  Supported: replaces, timer
  Remote-Party-ID: PBX-DID sip:PBX-DID@PROVIDER-IP
  ;party=calling;privacy=off;screen=no
  Content-Type: application/sdp
  Content-Length: 260
 
  v=0
  o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP
  s=Asterisk PBX 1.8.11.0
  c=IN IP4 PBX-PUBLIC_IP
  t=0 0
  m=audio 13042 RTP/AVP 18 101
  a=rtpmap:18 G729/8000
  a=fmtp:18 annexb=no
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-16
  a=ptime:20
  a=sendrecv
 
 
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Re: [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

[Sam Basan]

Hi,

It's looks like you are having NAT problem.
Packets from the provider fail reaching your box.

נשלח מטלפון נייד
בתאריך 14 באוג' 2015 15:56,‏ Daniel - Asterisk earohua...@gmail.com כתב:

 Hello friends:

 I am facing cutoffs randomly when negotiating calls.

 The PBX dials the destination, the provider (softswitch) receives the
 request *[1]* and sudenly the PBX hangs up the call* [2]* while the
 provider is still dialing it, as a consequence the remote peer receives a
 ghost call. Along the atempt I could see six times a messages regarding NAT
 isuues *[3]*

 I hope anyone can give me an idea to solve this issue. Softswitch is using
 an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with
 Asterisk 1.8.11.0

 Thanks in advance

 Elder D. Arohuanca
 Lima - Peru


 *[1]*
 [Aug 12 19:21:05] VERBOSE[17115] app_dial.c:-- Called
 SIP/SIP-PROVIDER/965034648


 *[2]*
 [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached
 on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno
 103 (Critical Request) -- See
 https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 8832ms with no response
 [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call
 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical
 packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 ).
 [Aug 12 19:21:14] VERBOSE[17115] app_dial.c:   == Everyone is
 busy/congested at this time (1:0/0/1)
 [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
 [s@macro-dialout-trunk:20] NoOp(SIP/143-01d8, Dial failed for some
 reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111) in new stack
 [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
 [s@macro-dialout-trunk:21] Goto(SIP/143-01d8, s-CHANUNAVAIL,1) in
 new stack

 *[3]*
 Retransmitting #3 (no NAT) to PROVIDER-IP:5060:
 INVITE sip:dialed_number@PROVIDER-IP SIP/2.0
 Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701
 Max-Forwards: 70
 From: PBX-DID sip:outbound-trunk@PROVIDER-IP;tag=as27ef83ae
 To: sip:dialed_number@PROVIDER-IP
 Contact: sip:outbound-trunk@PBX-PUBLIC_IP:5060
 Call-ID: 6b9ad82d4673fdab722f9e53411a767d@PROVIDER-IP
 CSeq: 103 INVITE
 User-Agent: FPBX-2.8.1(1.8.11.0)
 Proxy-Authorization: Digest username=outbound-trunk, realm=SoftSwitch,
 algorithm=MD5, uri=sip:dialed_number@PROVIDER-IP,
 nonce=d1b5806808a0888112190722408572932332,
 response=40c94f3c04e87e3382c7652d1f012dc9
 Date: Thu, 13 Aug 2015 00:56:40 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Remote-Party-ID: PBX-DID sip:PBX-DID@PROVIDER-IP
 ;party=calling;privacy=off;screen=no
 Content-Type: application/sdp
 Content-Length: 260

 v=0
 o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP
 s=Asterisk PBX 1.8.11.0
 c=IN IP4 PBX-PUBLIC_IP
 t=0 0
 m=audio 13042 RTP/AVP 18 101
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=no
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv


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Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Sam Basan]

Hi,

What you need is PRI TAP (also called SWITCH box)

Check this one
http://www.voicetronix.com/openpri.htm

נשלח מטלפון נייד
בתאריך 3 באוג' 2015 17:53,‏ Eric Klein eric.kl...@greenfieldtech.net
כתב:

 Hi all,

 Strange request, I have a customer where we are putting an Asterisk PBX in
 front of a legacy (non-VoIP) PBX. One of the requirements it that the
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards
 the carrier) with the ability to go to pass through should the Asterisk PBX
 (software or hardware level) fail.

 I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards.

 Does anyone know of a card that will do this? I know that Digium has an
 external box (the r850) that does something similar for 2 PBXs making them
 high availability, but in this case I only have the 1 Asterisk box acting
 as a gateway and passing some calls out over SIP and IAX2.

 Any suggestions would be appreciated.

 Thanks
 Eric

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Re: [asterisk-users] Outbound DID: in sip.conf or dialplan or db?

[Lutgring, Sam]

I have done this successfully in 2 ways depending on your requirements.  
Usually, I just set the callerid number right in the SIP, this is the easiest 
and cleanest in my opinion.  Worth mentioning that I always set the callerid in 
the SIP regardless, this way I know that internal calls, trunk calls, whatever 
are clean when it rings on any phone (including internal).  In this case my DID 
number (block of 100) are easily mapped to my extensions (forward planning 
during setup) because the extension is the last 4 digits of the DID.  This way 
in the default context I can say exten = _12345XX,1,Goto(default,${EXTEN:3},1) 
where the extension would be 45XX.

The second way that I have done this is by using the Asterisk internal database 
(no API or outside DB to worry about).  Then just simply do a DB lookup for the 
proper callerid before routing the call out.  Where I have had to use this is 
when multiple phones are grouped (multiple groups) to share a DID number for 
their callerid.

***
Sam Lutgring
Director of Informational Technology Services
Calhoun Intermediate school district
lutgr...@calhounisd.orgmailto:lutgr...@calhounisd.org
www.calhounisd.orghttp://www.calhounisd.org

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Tuesday, March 27, 2012 4:59 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Outbound DID: in sip.conf or dialplan or db?

I am setting up my dialplan with quite some outbound numbers. We have a block 
of 100 DID's, for which some of them will go direct to specific phones. I am 
struggling how to solve this, so I am searching for a little advice. These are 
my concerns.

I could set the DID in the sip.conf using something like:

callerid=137-Roland 31229253137

137 would be my extention number here.

I think the downside of this is, that I should configure this for each SIP 
account. I could specify a default callerid, which our main DID, in a template, 
but then people will see this general ID when I call internal extentions as 
well. This way the receiver cannot see my extention number.

Other solution would be to specify my DID in the dailplan. I tried this 
solution, which works:

exten = _00Z.,1,NoOp(Call Received from ${CALLERID(num)} to ${EXTEN:1})
exten = _00Z./Jeroen_S,2,Set(CALLERID(num)=31229700210)
exten = _00Z./Roland_odA,2,Set(CALLERID(num)=31852013900)
exten = _00Z./Schoolshopper,2,Set(CALLERID(num)=31852013900)
exten = _00Z.,2,Set(CALLERID(num)=31229700203)
  same = n,GotoIf($[${CALLERID(num)}=31852013900]?otconnect:voys)
  same = n(otconnect),Set(OUT=999210485)
  same = n,Goto(dodial)
  same = n(voys),Set(OUT=143810001)
  same = n(dodial),Dial(SIP/${EXTEN:1}@${OUT})

Upside is that I can be more specific in routing. The same handset could have 
different DID's, but I think usually the DID is bound to a SIP account. So I 
would probably create a second SIP account if a user needs an extra DID anyways.

The downside in my opinion is that my Dialplan will be filled with around 30 
extra lines with account specific stuff. I would rather keep my dialplan code 
clean. Also when I would have more patterns that can be matched, I have to 
specify them for this pattern as well. That would already take around 60 lines 
of extra code.

I have considered an AGI call to fetch the data from a database. But wouldn't 
this be a higher risk? When the database fails or is too slow, it will not 
work? I would rather use mysql than the asterisk db, because i can manage mysql 
easy with phpmyadmin.

Any suggestions would  be appreciated! Am I missing any options here?


This email is intended only for the use of the addressee(s) named herein. It 
may contain legally privileged and confidential information. If you are not the 
intended recipient, or an authorized representative of the intended recipient, 
you are hereby notified that any review, copying or distribution of this email 
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Re: [asterisk-users] How to add prefix in Extensions.Conf

[Lutgring, Sam]

The short answer is yes you can.  Now the longer answer is give us more detail 
if you want to know how.  Are they asking you to add the 92 when you dial 
5672531308, or is this question really about the callerid number?

***
Sam Lutgring
Director of Informational Technology Services
Calhoun Intermediate school district
lutgr...@calhounisd.orgmailto:lutgr...@calhounisd.org
www.calhounisd.orghttp://www.calhounisd.org

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Ali
Sent: Friday, March 23, 2012 5:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to add prefix in Extensions.Conf


Hello,


I have a DID number 5672531308 , I want to add 92 prefix in it as been told by 
my provider , so I can I do this in extensions.conf?
--
Regards,

Muhammad Ali
DIDx SUPPORT
http://whttp://w/ww.didx.nethttp://ww.didx.net/
Skype: didxnet
Phone: +1-212-655-5763 / +1-850-433-8555
Direct : +1-567-2531308
http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded





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this email from your system. Thank you.
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Re: [asterisk-users] Phone Inventory

[Muro, Sam]

Thank you all

You are a life saver

Sam

Dale Noll wrote:
 On 02/23/2012 08:49 AM, Danny Nicholas wrote:
 Here is a snippet that somebody smarter than I am can improve upon
 for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
 sip
 show peer $a;done|grep Useragent
 for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
 sip
 show peer $a;done|grep Contact


 Thanks for the inspiration!!

 Here is my version, done with a single loop and gets Useragent and
 Contact together with a visual separation between peers.


 asterisk -rx sip show peers|
 cut -f1 -d/ | grep -P '\d\d\d\d' |
 grep -vP '(UNKNOWN|Unmonitored)' |
 while read PEER
 do
 asterisk -rx sip show peer ${PEER} |
 grep -P (Useragent|Contact)
 echo 
 done

 I hope others find it useful.

 Dale

 PS. I by no means claim to be smarter than thou.  I just happen to
 really like grep and the -P option  ;-)

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[asterisk-users] Replicating SIP registration Info between active to standby

[Muro, Sam]

I have a scenario whereby two servers are acting in active-standby mode.
In case the active server fail, the shared IP is activated on standby
server for continuity.

However, SIP phones (all are Polycom) takes quite a long time to register
to the Standby Server (up to 1-10min). While Polycom allow double
registration, we would like to make it simple by provision only one
registration server at a time.

How can I copy sip registration information from Active Server to Standby
Server

Sam


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Re: [asterisk-users] Replicating SIP registration Info between active to standby

[Muro, Sam]

Hi Takehiro

Are you suggesting sharing the AstDB ?

Sam

Takehiro Matsushima wrote:
 Hi.

 How about place backend DB on shared disk, or make replication between
 them?
  2012/02/24 13:58 Muro, Sam resea...@businesstz.com:

 I have a scenario whereby two servers are acting in active-standby mode.
 In case the active server fail, the shared IP is activated on standby
 server for continuity.

 However, SIP phones (all are Polycom) takes quite a long time to
 register
 to the Standby Server (up to 1-10min). While Polycom allow double
 registration, we would like to make it simple by provision only one
 registration server at a time.

 How can I copy sip registration information from Active Server to
 Standby
 Server

 Sam


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[asterisk-users] Phone Inventory

[Muro, Sam]

Hi there

I have just took a support of a customer with hundreds of IP phones,
mostly Polycom with mixed models.

Is there a way to query asterisk or any other command to retrieve the
inventory of all connected phones. i.e. Phone Type and Phone Model, say
Polycom SPIP331 or so

Thanks
Sam

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[asterisk-users] Asterisk Security: Allow only one phone per sip registration

[Muro, Sam]

Hi there

Consider this. You have three SIP extension 200, 201 and 202 and you have
configured your phones, say Polycom 331 to those accounts. 200 being one
very sensitive individual.

Lets say, an insider, get a new phone or perhaps an xlite and configure it
with the same extension, 200. Asterisk will register it as 200 to the new
IP address.  Now extension 202 call 200. The hacker answers it and pretend
is the same person. Do what he want to do and thats it.

Question;
How can i stop this type of threat

Regads
Peter

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Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

[Muro, Sam]

Terry Wilson wrote:
 - Original Message -
 From: Sam Muro resea...@businesstz.com
 To: asterisk-users@lists.digium.com
 Sent: Friday, October 14, 2011 2:02:01 AM
 Subject: [asterisk-users] Asterisk Security: Allow only one phone per
 sip registration
 Hi there

 Consider this. You have three SIP extension 200, 201 and 202 and you
 have
 configured your phones, say Polycom 331 to those accounts. 200 being
 one
 very sensitive individual.

 Lets say, an insider, get a new phone or perhaps an xlite and
 configure it
 with the same extension, 200. Asterisk will register it as 200 to the
 new
 IP address. Now extension 202 call 200. The hacker answers it and
 pretend
 is the same person. Do what he want to do and thats it.

 Question;
 How can i stop this type of threat

 I would recommend actually setting a different secret field in sip.conf
 for each device so that your would-be attacker isn't able to register as
 someone else.

Is there a way one can bind sip account to specific mac-address (assume on
the same subnet). In this way, even if you know the username/secret, you
will still have to use the same physical phone, unless you play with
mac-address.

 Or you could buy a gun. I bet the insider would be very
 afraid of the gun and would therefore avoid any shenanigans while you were
 around. This would especially be true if you randomly shot items like
 coffee cups and plants whenever you thought they were looking at you
 funny. That'll show 'em.

Lol! Here they will name you a terrorist


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Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

[Muro, Sam]

Terry Wilson wrote:

 Is there a way one can bind sip account to specific mac-address
 (assume on
 the same subnet). In this way, even if you know the username/secret,
 you
 will still have to use the same physical phone, unless you play with
 mac-address.

 No. And mac addresses are easily spoofed so it would not help. Use
 passwords. Keep them safe.

Thanks. Let me see how best i can complicate them per phone. Ooops, 1000
sip phones


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Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

[Muro, Sam]

Thanks Terry!
Let me think of all possibilities and shall holla. Can you be one?


Terry Wilson wrote:
 Thanks. Let me see how best i can complicate them per phone. Ooops,
 1000
 sip phones

 If it were me, I would look into Asterisk Realtime for handling the SIP
 phones. I would then write a script to generate the configs for the phones
 into the SIP realtime database with random passwords. Match up the phones
 with the accounts and provision the phones. You would most likely use a
 provisioning server of some kind to generate the actual phone
 configurations. You can check out the res_phoneprov module in Asterisk,
 find another one somewhere, or write your own. Many people tend to write
 their own for large installations. I did.

 If you have a big installation like this and are wondering about things
 like whether mac addresses should be used for security, it might also be a
 good idea to hire a consultant. Check out the asterisk-biz mailing list.

 Terry

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Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

[Muro, Sam]

Thanks A.J

I know and I can assure you no one will get that physical access to the
system.

A J Stiles wrote:
 On Friday 14 October 2011, Muro, Sam wrote:
 Hi there

 Consider this. You have three SIP extension 200, 201 and 202 and you
 have
 configured your phones, say Polycom 331 to those accounts. 200 being one
 very sensitive individual.

 Lets say, an insider, get a new phone or perhaps an xlite and configure
 it
 with the same extension, 200. Asterisk will register it as 200 to the
 new
 IP address.  Now extension 202 call 200. The hacker answers it and
 pretend
 is the same person. Do what he want to do and thats it.

 Question;
 How can i stop this type of threat

 Be careful who you employ and how you treat them  :)

 Once someone has physical access to your equipment, all bets are off .

 --
 AJS

 Answers come *after* questions.

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Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

[Sam Govind]

Hey,
Why do you think using OpenSIPs is not going to work for you ? You can
always add SIP trunks on openSips and based upon which trunks getting the
call you can LB or/and FO to as many asterisk servers as you want !

Regards,
-Sammy

On Sun, Oct 2, 2011 at 7:12 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  If one server is supposed to carry the full load of the other during
 failure, then you have to size each server to handle  100% load - so load
 balancing is pointless.

 Checkout haast at www.generationd.com and read  the docs on how it does
 failover...certainly good for ideas.

  --
 *From:* asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Tobias Steen [
 tobias.st...@s2.se]
 *Sent:* Saturday, October 01, 2011 6:30 PM
 *To:* Asterisk Users List
 *Subject:* [asterisk-users] Make asterisk cluster appear and operate as a
 single server?

   Hi,



 I'm trying to plan a system of clustered asterisk machines where a number
 of SIP trunks will be hosted on the platform. Each trunk will be hosted for
 a specific customer who owns it and therefore payment is handled directly
 between the customers and their trunk-providers, each trunk will have about
 50-200 simultaneous calls.



 No SIP phones will be directly connected to the platform, my thought is
 that the asterisk machines should only receive incoming and make outgoing
 calls through the trunks, and then connect the calls with each other.



 To make this scalable and have the option of running an infinite number of
 sip-trunks, I need a good way to load-balance my asterisk servers and
 implement failover support and also be able to add / replace the machines in
 the cluster in a safe and reliable way.



 I'm have some experience building single asterisk solutions but I have
 never worked with load balancing of multiple asterisk machines.



 Is it possible to configure all trunks on a single asterisk setup which is
 then reflected over a cluster of asterisk machines? If I have a cluster of
 machines, I guess I need some kind of front-end application / system? I will
 then also need to be able to connect calls between the machines, the calls
 to be connected with each other will always be incoming and outgoing on the
 same trunk.



 In other words, I want to create a large cluster of asterisk machines to
 appear and operate as a single asterisk server.



 I've looked at projects like OpenSIP but it feels like this is not really
 what I need?



 I really appreciate if someone can help me get on the correct path here, I
 need all the feedback I can get.





 Thanks in advance!





 Best regards

 Tobias



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Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

Though its in Spanish - Hope it is exactly what OP requires !

http://www.voztovoice.org/?q=node/477

Found it via : http://www.voip-info.org/ News

Regards,
-Sammy

On Thu, Sep 22, 2011 at 6:50 PM, bakko asannu...@gmail.com wrote:

 Hi look at option a. This option put on accountcode field the name on the
 left your password file.

 Regards

 Enviado desde mi iPad

 El 22/09/2011, a las 5:49, Malvin Rito mr...@mail.altcladding.com.ph
 escribió:

 Hi,

 I tried Authenticate where pass codes are stored on the file pass.conf and
 it works.

 exten = _,1,Authenticate(/etc/asterisk/pass.conf)

 Since I have my CDR, I want to have a report wherein I can check which pass
 code did the call. How can I achieve it?
 Using authenticate through file does not replace ACCOUNT_CODE field with
 the pass code entered, it only show *ast_h323 *under the Account_Code
 field.

 Regards,
 Malvin

 On 9/21/2011 1:01 PM, Sam Govind wrote:

 See core show application autheTAB
 If passwords are already the same as those of voicemail.conf go for
 application VMAuthenticate() - DIA generates a dial-tone which I don't think
 is suitable for dialling out from users(insiders)

-= Info about application 'Authenticate' =-

  [Synopsis]
 Authenticate a user

  [Description]
 This application asks the caller to enter a given password in order to
 continue
 dialplan execution.
 If the password begins with the '/' character,  it is interpreted as a file
 which contains a list of valid passwords, listed 1 password per line in the
 file.
 When using a database key, the value associated with the key can be
 anything.
 Users have three attempts to authenticate before the channel is hung
 up.

  [Syntax]
 Authenticate(password[,options[,maxdigits[,prompt]]])

  [Arguments]
 password
 Password the user should know
 options
 a: Set the channels' account code to the password that is entered
 d: Interpret the given path as database key, not a literal file
 m: Interpret the given path as a file which contains a list of account
 codes and password hashes delimited with ':', listed one per line in
 the
 file. When one of the passwords is matched, the channel will have its
 account code set to the corresponding account code in the file.
 r: Remove the database key upon successful entry (valid with 'd'
 only)
 maxdigits
 maximum acceptable number of digits. Stops reading after maxdigits
 have been entered (without requiring the user to press the '#' key).
 Defaults to 0 - no limit - wait for the user press the '#' key.
 prompt
 Override the agent-pass prompt file.

  [See Also]
 VMAuthenticate(), DISA()


  On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito mr...@mail.altcladding.com.ph
 mr...@mail.altcladding.com.ph wrote:

  Thanks. ?If I want to use unique PIN for every user that dials out how
 can I implement it using Authenticate app?

 Regards,
  Malvin


 On 9/21/2011 12:42 PM, Sam Govind wrote:

 DISA and DB based Auth could be an overkill.

 Kyle showed the very simplistic dial plan if Dial-out pin is common for
 the whole system.
 See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
 if Voicemail PIN are required to be used use application 
 *MAuthenticate([mailbox][@context][,options]
 *

  Regards,

  - Sammy

 On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton  k...@mocker.org
 k...@mocker.org wrote:

 Something like this should work:

  exten = _011.,1,Answer
 exten = _011.,n,Wait(1)
 exten = _011.,n,Read(password,enter-password,5)
 exten = _011.,n,GotoIf($[${password} = 12345]?5:9)

  exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
 exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

  exten = _011.,n,Hangup
 exten = _011.,n,Playback(invalid)
 exten = _011.,n,Hangup

  Could be cleaned up (the GotoIf isn't very descriptive about where it's
 going), but it's a starting point.


  On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:

  Hi List,
 I currently have a asterisk server running used for dialing-out for IDD
 but I want to Put a pincode wherein only users with the right pin code will
 be allowed to dial IDD. Any sample dialplan you can suggest pls?

 Thanks,
 Malvin
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Re: [asterisk-users] invite authentication error !?

[Sam Govind]

What Sip declaration are you using for the remote sip proxy in sip.conf?

On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov
abalas...@evaristesys.comwrote:

 This is just a speculative shot in the dark, but remember that the domain
 of the From URI is important, and that the authentication realm (domain)
 is part of the authentication credentials.  So, what you have in your
 'fromdomain' and 'host' settings on the peer does matter.

 --
 This message was painstakingly thumbed out on my mobile, so apologies for
 brevity, errors, and general sloppiness.

 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/http://www.evaristesys.com/

 On Sep 30, 2011, at 3:16 AM, cnasterisk cnaster...@163.com wrote:

 hi,
Dear all.
I setted a sip account on a sip trunk. when a  client call via this sip
 trunk, asterisk call failed on this trunk.
 I have captured the sip messages on the host where asterisk located, and
 found that:

 1. asterisk send a INVITE message to remote sip proxy without
 proxy-authorization field.
 2. the remote sip proxy send back a
  SIP/2.0 407 Proxy Authentication Required message.
 3. asterisk send a INVITE message with  proxy-authorization field.
 4. remote proxy send back a 403(Forbidden) message， that is mean wrong
 password

 I also tested the sip account on a softphone, it works normal!

 why this happed？ and how can i solve it?



 2011-09-30
 --
 kevin

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Re: [asterisk-users] invite authentication error !?

[Sam Govind]

Whatever that remote party is, you are most definitely using
a username/secret declaration for that. So the sip attributes set for that
proxy define the behaviour for this.

On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk cnaster...@163.com wrote:

 **
  hi, Sam
 thanks for your kindly reply.
 The remote proxy is not asterisk

 2011-09-30
 --
  cnasterisk
 --
 *发件人：* Sam Govind
 *发送时间：* 2011-09-30  15:36:41
 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
 *抄送：*
 *主题：* Re: [asterisk-users] invite authentication error !?
  What Sip declaration are you using for the remote sip proxy in sip.conf?

 On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov abalas...@evaristesys.com
  wrote:

  This is just a speculative shot in the dark, but remember that the
 domain of the From URI is important, and that the authentication realm
 (domain) is part of the authentication credentials.  So, what you have in
 your 'fromdomain' and 'host' settings on the peer does matter.

 --
 This message was painstakingly thumbed out on my mobile, so apologies for
 brevity, errors, and general sloppiness.

 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/http://www.evaristesys.com/

 On Sep 30, 2011, at 3:16 AM, cnasterisk cnaster...@163.com wrote:

   hi,
Dear all.
I setted a sip account on a sip trunk. when a  client call via this sip
 trunk, asterisk call failed on this trunk.
 I have captured the sip messages on the host where asterisk located, and
 found that:

 1. asterisk send a INVITE message to remote sip proxy without
 proxy-authorization field.
 2. the remote sip proxy send back a
  SIP/2.0 407 Proxy Authentication Required message.
 3. asterisk send a INVITE message with  proxy-authorization field.
 4. remote proxy send back a 403(Forbidden) message， that is mean wrong
 password

 I also tested the sip account on a softphone, it works normal!

 why this happed？ and how can i solve it?



 2011-09-30
 --
 kevin

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Re: [asterisk-users] invite authentication error !?

[Sam Govind]

Please post some configurations. YOU CAN REMOVE username / secret from the
sip configs but posting remaining configs will only help to your issue
resolution, Alex first words were This is just a speculative shot in the
dark .

Fine if you were just Trolling here !


2011/9/30 cnasterisk cnaster...@163.com

 **
 asterisk can register successfully on the remote party. so i think username
  password must be ok


 2011-09-30
 --
  cnasterisk
 --
 *发件人：* Sam Govind
 *发送时间：* 2011-09-30  16:05:21
 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
 *抄送：*
 *主题：* Re: [asterisk-users] invite authentication error !?
  Whatever that remote party is, you are most definitely using
 a username/secret declaration for that. So the sip attributes set for that
 proxy define the behaviour for this.

 On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk cnaster...@163.com wrote:

 **
  hi, Sam
  thanks for your kindly reply.
 The remote proxy is not asterisk

 2011-09-30
 --
  cnasterisk
 --
 *发件人：* Sam Govind
 *发送时间：* 2011-09-30  15:36:41
  *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
 *抄送：*
  *主题：* Re: [asterisk-users] invite authentication error !?
   What Sip declaration are you using for the remote sip proxy in
 sip.conf?

 On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov 
 abalas...@evaristesys.com wrote:

  This is just a speculative shot in the dark, but remember that the
 domain of the From URI is important, and that the authentication realm
 (domain) is part of the authentication credentials.  So, what you have in
 your 'fromdomain' and 'host' settings on the peer does matter.

 --
 This message was painstakingly thumbed out on my mobile, so apologies for
 brevity, errors, and general sloppiness.

 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/http://www.evaristesys.com/

 On Sep 30, 2011, at 3:16 AM, cnasterisk cnaster...@163.com wrote:

   hi,
Dear all.
I setted a sip account on a sip trunk. when a  client call via this
 sip trunk, asterisk call failed on this trunk.
 I have captured the sip messages on the host where asterisk located, and
 found that:

 1. asterisk send a INVITE message to remote sip proxy without
 proxy-authorization field.
 2. the remote sip proxy send back a
  SIP/2.0 407 Proxy Authentication Required message.
 3. asterisk send a INVITE message with  proxy-authorization field.
 4. remote proxy send back a 403(Forbidden) message， that is mean wrong
 password

 I also tested the sip account on a softphone, it works normal!

 why this happed？ and how can i solve it?



 2011-09-30
 --
 kevin

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Re: [asterisk-users] PSTN connectivity

[Sam Govind]

The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
there is some misconfiguration in FreePBX and your dialled number is not
hitting any dial-able rule.  See your FreePBX guide.


On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote:

 Hi,

   Please see the sample.

 A ) Analog HardwareType Ports Action   FXO Ports 1 
 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo
   FXS
 Ports --

 B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog*

 *
 C ) ZAP Trunk (DAHDI compatibility Mode)*


 Trunk Description:
 Outbound Caller ID:CID Options:
   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
 Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
   Dial Rules Wizards:
   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk name):



 *D ) INBOUND route *

  Description:
 Extensions: 199
 *

 E ) **OUTBOUND Route*

 Route Name:  9_outside  Route CID:  Override Extension CID  Route
 Password:  PIN Set:
  Emergency Dialing:  Intra Company Route:  Music On Hold?
   Dial Patterns
 8|NXXNXX 8|NXX
   Dial patterns wizards*: *
   Trunk SequenceZAP/g0  0
 *
 F ) In command Line I can see the following things *


 [root@astrisks ~]# *dahdi_cfg -vv*


 DAHDI Tools Version - 2.3.0

 DAHDI Version: 2.3.0.1
 Echo Canceller(s):
 Configuration
 ==


 Channel map:

 Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)

 1 channels to configure.

 Setting echocan for channel 1 to none


 [root@astrisks ~]# *dahdi_scan*

 [1]
 active=yes
 alarms=OK
 description=Wildcard X100P Board 1
 name=WCFXO/0
 manufacturer=Digium
 devicetype=Wildcard X100P
 location=PCI Bus 02 Slot 02
 basechan=1
 totchans=1
 irq=193
 type=analog
 port=1,FXO



 *Asterisk CLI*


 *astrisks*CLI dahdi show status*

 Description  Alarms  IRQbpviol CRC4   Fra
 Codi Options  LBO
 Wildcard X100P Board 1   OK  0  0  0  CAS
 Unk   0 db (CSU)/0-133 feet (DSX-1)

 *
 output when i dialing to a local number*

 Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
 Verbosity is at least 3
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Executing [s@from-internal:1] Macro(SIP/199-003a,
 hangupcall) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
 -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in
 new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'SIP/199-003a' in macro 'hangupcall'
   == Spawn extension (from-internal, s, 1) exited non-zero on
 'SIP/199-003a'
 -- Executing [h@from-internal:1] Macro(SIP/199-003a,
 hangupcall) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in
 new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'SIP/199-003a' in macro 'hangupcall'
   == Spawn extension (from-internal, h, 1) exited non-zero on
 'SIP/199-003a'

















 On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote:

 Some CLI logs will get you better help on the issue ! also paste the FXO
 configurations and how you configured it !

 On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote:

 Hi All,

   I am trying to connect my asterisk box with freepbx to PSTN. I
 have purchased x100p FXO card and installed in my asterisk server. My
 freepbx detected the x100p FXO card and i can see the card specific details
 in command line. I have configured the following things.

 1. OUTBOUND caller id and Dialing rules in Freepbx.

 2. INBOUND route

 When i call to the PSTN number before connecting to the FXO card, i am
 getting a ringing. But i get a message like the number is out of order
 when i just connect the line to FXO card.

 Please some one help me to resolve his issue

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Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Actually its easier. I haven't worked on FreePBX lately so what I remember
is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep
it empty as well. Then you've created an outbound route its dial-rule is
important.

But the funny thing which I didn't mention before is that you've ZAP defined
in FreePBX but actually its DAHDI so I remember they've this cute parameter
in amportal.conf which tells FreePBX to convert ZAP into DAHDI.



On Thu, Sep 29, 2011 at 11:57 AM, michael k mich...@inapp.com wrote:

 Can you please figure out the configuration issue in my freepbx ?





 On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote:

 The Call at this point is not even looking for FXO/Dahdi/Zap.. See the
 CLI. there is some misconfiguration in FreePBX and your dialled number is
 not hitting any dial-able rule.  See your FreePBX guide.


 On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote:

 Hi,

   Please see the sample.

 A ) Analog HardwareType Ports Action   FXO Ports 1 
 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo
   FXS
 Ports --

 B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: *from-analog
 *

 *
 C ) ZAP Trunk (DAHDI compatibility Mode)*


 Trunk Description:
 Outbound Caller ID:CID Options:
   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
 Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
   Dial Rules Wizards:
   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk
 name):


 *D ) INBOUND route *

  Description:
 Extensions: 199
 *

 E ) **OUTBOUND Route*

 Route Name:  9_outside  Route CID:  Override Extension CID  Route
 Password:  PIN Set:
  Emergency Dialing:  Intra Company Route:  Music On Hold?
   Dial Patterns
 8|NXXNXX 8|NXX
   Dial patterns wizards*: *
   Trunk SequenceZAP/g0  0
 *
 F ) In command Line I can see the following things *


 [root@astrisks ~]# *dahdi_cfg -vv*


 DAHDI Tools Version - 2.3.0

 DAHDI Version: 2.3.0.1
 Echo Canceller(s):
 Configuration
 ==


 Channel map:

 Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)

 1 channels to configure.

 Setting echocan for channel 1 to none


 [root@astrisks ~]# *dahdi_scan*

 [1]
 active=yes
 alarms=OK
 description=Wildcard X100P Board 1
 name=WCFXO/0
 manufacturer=Digium
 devicetype=Wildcard X100P
 location=PCI Bus 02 Slot 02
 basechan=1
 totchans=1
 irq=193
 type=analog
 port=1,FXO



 *Asterisk CLI*


 *astrisks*CLI dahdi show status*

 Description  Alarms  IRQbpviol CRC4   Fra
 Codi Options  LBO
 Wildcard X100P Board 1   OK  0  0  0  CAS
 Unk   0 db (CSU)/0-133 feet (DSX-1)

 *
 output when i dialing to a local number*

 Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
 Verbosity is at least 3
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Executing [s@from-internal:1] Macro(SIP/199-003a,
 hangupcall) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
 -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, )
 in new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'SIP/199-003a' in macro 'hangupcall'
   == Spawn extension (from-internal, s, 1) exited non-zero on
 'SIP/199-003a'
 -- Executing [h@from-internal:1] Macro(SIP/199-003a,
 hangupcall) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in
 new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'SIP/199-003a' in macro 'hangupcall'
   == Spawn extension (from-internal, h, 1) exited non-zero on
 'SIP/199-003a'

















 On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote:

 Some CLI logs will get you better help on the issue ! also paste the FXO
 configurations and how you configured it !

 On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote:

 Hi All,

   I am trying to connect my asterisk box with freepbx to PSTN.
 I have purchased x100p FXO card and installed in my asterisk server. My
 freepbx detected the x100p FXO card and i can see the card specific 
 details

Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Hey Warren I thought that these are the complete CLI logs for one call. It
started like   == Using SIP RTP CoS mark 5 and from-internal priority-1
..So that seemed legit to me. Yeah I too suspect that dialing rules are not
being matched and thats why Gotoif's are failing.

On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby wcse...@selbytech.com wrote:


 On Thu, Sep 29, 2011 at 7:51 AM, michael k mich...@inapp.com wrote:

 Thanks for the update. but how do i resolve this issue ? can you help me
 please ?


 You didn't provide a full CLI trace of the outgoing call, you only supplied
 the hangup portion of the call.  Please try again.

 Also, what are the dialing rules like in your country?  You only have
 outbound dial patterns setup to handle North American numbers (8+ NXXNXX
 or 8+ NXX).
 The Dial Pattern box in the Outbound Rules box is where you define what
 numbers you want to go out over this trunk.  If you dial a number that
 doesn't match one of these
 patterns, FreePBX is going to look internally for a dial pattern to match
 against, and if it doesn't find one there, it will end the call.


 --
 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com


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Re: [asterisk-users] record calls of specific agnets

[Sam Govind]

I guess that was this variable like SPYGROUP which needs to be set for
specific extensions and then ask Chanspy to spy on that group. !!

On Thu, Sep 29, 2011 at 9:37 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:

 On Thursday 29 September 2011, Lyle McKarns wrote:
  Hello Asterisk List!
 
  I have been asked to record calls from specific agents, and I am having
  difficulty finding if this is possible, and if so, how exactly to do it.

 Have a recorded context and an unrecorded context in your dialplan,
 identical save for the lines that start the recording and cleanup processes
 being absent from the latter.  Then set contexts per extension in sip.conf.

 --
 AJS

 Answers come *after* questions.

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Re: [asterisk-users] Features not working

[Sam Govind]

Hey,
Whats the output of command features show ? on CLI ?


On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl mdi...@diehlnet.com wrote:

 Hi all.

 I could have sworn this working at one time...

 But it doesn't look like any of the functions provided by features.so is
 working for me.  (one-touch monitoring, attended/blind transfer, etc)

 I've (re)loaded features.so, as well as bridge_builtin_features.so.

 The config file looks sane.

 What else should I try?

 TIA,

 --

 Take care and have fun,
 Mike Diehl.

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Re: [asterisk-users] Call does not pass through

[Sam Govind]

Hey,

So far the Dialplan execution is ok, despite the conflicts and some other
mistakes like repeating priorities in it but they're not involved in this
call.

You'r A-leg is H323 endpoint and Destination is on SIP. I'm now thinking
about codec mismatch on first try Tell me this happens every time? Like
first call fails for sure and second call goes through?

if so please post the tcpdump/wireshark traces for a failed call as well as
successful call separately. Also NAting could be the reason like Rube
suspects - but the call should be failing everytime. Anyways post the
complete h323 as well as SIP traces combined for each failed  successful
call.

Regards.
-Sammy

On Wed, Sep 28, 2011 at 10:59 AM, Malvin Rito mr...@mail.altcladding.com.ph
 wrote:

  Thanks Sam. Please see below CLI log:

 *[root@localhost ~]# asterisk -r
 Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
 Created by Mark Spencer marks...@digium.com marks...@digium.com
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
 detail

 s.
 This is free software, with components licensed under the GNU General
 Public
 License version 2 and other licenses; you are welcome to redistribute it
 under
 certain conditions. Type 'core show license' for details.
 =
   == Parsing '/etc/asterisk/asterisk.conf':   == Found
 Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
 Verbosity is at least 4
   == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on
 'OOH323

 /(null)-b7798910'
 -- Executing [s@avaya-internal:1] Answer(OOH323/(null)-b7798910, )
 in ne

 w stack
 -- Executing [s@avaya-internal:2] BackGround(OOH323/(null)-b7798910,
 pls-

 entr-num-uwish2-call) in new stack
 -- OOH323/(null)-b7798910 Playing 'pls-entr-num-uwish2-call.gsm'
 (language

 'en')
   == CDR updated on OOH323/(null)-b7798910
 -- Executing [15707088788@avaya-internal:1]
 Authenticate(OOH323/(null)-b779


 8910, /etc/asterisk/passcode.txt,a) in new stack
 -- OOH323/(null)-b7798910 Playing 'agent-pass.ulaw' (language 'en')
 -- OOH323/(null)-b7798910 Playing 'auth-thankyou.ulaw' (language
 'en')
 -- Executing [15707088788@avaya-internal:2]
 Dial(OOH323/(null)-b7798910, 


 SIP/15707088788@cordia) in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called 15707088788@cordia
 -- SIP/cordia-0017 answered OOH323/(null)-b7798910
   == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on
 'OOH323

 /(null)-b7798910'
 -- Executing [s@avaya-internal:1] Answer(OOH323/(null)-0a389388, )
 in ne

 w stack
 -- Executing [s@avaya-internal:2] BackGround(OOH323/(null)-0a389388,
 pls-

 entr-num-uwish2-call) in new stack
 -- OOH323/(null)-0a389388 Playing 'pls-entr-num-uwish2-call.gsm'
 (language

 'en')
 -- Executing [s@avaya-internal:3] WaitExten(OOH323/(null)-0a389388,
 ) in

 new stack
   == CDR updated on OOH323/(null)-0a389388
 -- Executing [18772281023@avaya-internal:1]
 Authenticate(OOH323/(null)-0a38


 9388, /etc/asterisk/passcode.txt,a) in new stack
 -- OOH323/(null)-0a389388 Playing 'agent-pass.ulaw' (language 'en')
 -- OOH323/(null)-0a389388 Playing 'auth-thankyou.ulaw' (language
 'en')
 -- Executing [18772281023@avaya-internal:2]
 Dial(OOH323/(null)-0a389388, 


 SIP/18772281023@cordia) in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called 18772281023@cordia
 -- SIP/cordia-0018 is making progress passing it to
 OOH323/(null)-0a3893

 88
 -- SIP/cordia-0018 is ringing
 -- SIP/cordia-0018 is making progress passing it to
 OOH323/(null)-0a3893

 88
 -- SIP/cordia-0018 answered OOH323/(null)-0a389388
   == Spawn extension (avaya-internal, 18772281023, 2) exited non-zero on
 'OOH323

 /(null)-0a389388'
   == Manager 'admin' logged on from 127.0.0.1
   == Manager 'admin' logged off from 127.0.0.1
   == Manager 'admin' logged on from 127.0.0.1
   == Manager 'admin' logged off from 127.0.0.1
 localhost*CLI
 Disconnected from Asterisk server
 Executing last minute cleanups
 [root@localhost ~]# nano /etc/asterisk/extensions_custom.conf
 [root@localhost ~]# asterisk -r
 Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
 Created by Mark Spencer marks...@digium.com marks...@digium.com
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
 details.
 This is free software, with components licensed under the GNU General
 Public
 License version 2 and other licenses; you are welcome to redistribute it
 under
 certain conditions. Type 'core show license' for details.
 =
   == Parsing '/etc/asterisk/asterisk.conf':   == Found
 Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
 Verbosity is at least 4
 -- Remote UNIX

Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Some CLI logs will get you better help on the issue ! also paste the FXO
configurations and how you configured it !

On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote:

 Hi All,

   I am trying to connect my asterisk box with freepbx to PSTN. I
 have purchased x100p FXO card and installed in my asterisk server. My
 freepbx detected the x100p FXO card and i can see the card specific details
 in command line. I have configured the following things.

 1. OUTBOUND caller id and Dialing rules in Freepbx.

 2. INBOUND route

 When i call to the PSTN number before connecting to the FXO card, i am
 getting a ringing. But i get a message like the number is out of order
 when i just connect the line to FXO card.

 Please some one help me to resolve his issue

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Re: [asterisk-users] number of calls simultaneous from AMI

[Sam Govind]

If you can post any relevant code sections and CLI output for this then
it'll be lot better to determine whats causing this. I never got any problem
initiating as many call as u can say from AMI !

On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis ge...@pagestation.com wrote:

  I am starting up 4 calls over the AMI.
 It appears as though the first 3 start up and go out right away.
 The 4th call is delayed like 15 seconds.

 Any thoughts on why this fourth call might be getting delayed...

 Everything is working its just delayed.

 Jerry

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Correct me if I'm wrong or don't know anything other than AMI Originate
Event or a call file to kick start a call from asterisk ! So making a new or
modifying asterisk call-file cron job/poller seems like a nice idea but why
put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
The simple idea is create a MySQL trigger for your Table insertion, the data
in the table at insertion time becomes parameters for a simple script that
triggers an AMI event (or call file) whichever is easier for you.

On Tue, Sep 27, 2011 at 6:54 PM, Nick Khamis sym...@gmail.com wrote:

 Hello David,

 At first I assumed asterisk used call files out of the box for
 normal-initiated/instantiated calls however,
 this is incorrect. I think call files was the easy approach for client
 just to place a file with call details
 in some location. I am trying to do the same with a db record. My
 first question is, how does asterisk
 initiate calls, i.e. what part of the source code is responsible for
 that. Are there any threads involved etc.

 Cheers,

 Nick.

 On Tue, Sep 27, 2011 at 9:35 AM, David Moring dmor...@tmcentral.net
 wrote:
  Hi Nick,
 
  Understand your reasoning - though as Matt points out sql db isn't in the
  core so compiling it there would preclude seemless upgrades.  Also, I
  personally would be concerned putting the calls right into the call-file
  thread might create an issue if you hung on a db query or insert.
  Finally
  (and I'd love to hear the answer not knowing), but I believe
  normally-initiated/instantiated calls are handled with direct calls via
  either SIP requests and/or AMI - thus even using the proposed method, I
  *think* the db/file-drop method is going to create some overhead that
 might
  not scale well...
 
  Best,
 
  David
 
  -Original Message-
  From: Nick Khamis sym...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Date: Mon, 26 Sep 2011 18:49:07 -0400
  Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
 
  Hello David,
 
  Thank you so much for your response. I am sure it can be easily done
  using AGI. The reason I am leaning more
  towards storing the call information in a database record, is because
  our existing client applications can be easily
  modified to write to MySQL. The asterisk cron/thread that would
  querying the DB should be no different than existing implementation
  used process the call files?
  For those of you that may be interested in what we are doing. We are
  developing an application that will apply NLP
  services on text generated using the speech to text module, and
  generate the response that will then be forwarded to
  the text to speech.
 
  Cheers,
 
  Nick
  .
 
 
 
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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

[Sam Govind]

:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue variables are not populated by then so filename won't contain the
Agent Number.
UNLESS you move the file after queue to a new filename containing the Agent
Number.

like ;

exten = whatever,n,SET(MONITOR_FILENAME=blah-blah)
exten = whatever,n,Queue(${params}); Queue should contain option c to
continue in dialplan when callee hangup. Caller hangup case needs special
attention too
exten = whatever,n,System(mv ${old-Filename} ${old-Filename}-
${MEMBERINTERFACE})

I guess this should do the job.

On Tue, Sep 27, 2011 at 8:30 PM, Carlos Chavez cur...@telecomabmex.comwrote:

 On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
  Dears;
 
  I am facing now a problem in the recording the calls that coming via the
 queue, the problem that I am not able to make the filename contains the
 agent (for example its extension) who received the call.
 
  Actually by looking to the below settings, it is clear that the agent
 name (it the phone extension or it is sip username .. etc) will not be
 included in the filename.
 
  How can I include the agent name in the filename? Because in outboud it
 is easy as the ${CHANNEL} will contain the sip username of the IP Phone but
 in the outbound it will contain the DAHDI channel that the call came via it
 .. so How to inlude the sip username for the IP Phone of the agent that is
 going to get the call from the queue?
 
  exten =
 s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
  exten = s,2,Queue(OrangeCMG,t,,,180)
  exten = s,3,Macro(voicemail,SIP/reception)
 
  Regards
  Bilal
 
 
 ; If set to yes, just prior to the caller being bridged with a queue
 member
 ; the following variables will be set
 ; MEMBERINTERFACE is the interface name (eg. Agent/1234)
 ; MEMBERNAME is the member name (eg. Joe Soap)
 ; MEMBERCALLS is the number of calls that interface has taken,
 ; MEMBERLASTCALL is the last time the member took a call.
 ; MEMBERPENALTY is the penalty of the member
 ; MEMBERDYNAMIC indicates if a member is dynamic or not
 ; MEMBERREALTIME indicates if a member is realtime or not
 ;
 ;setinterfacevar=no

 Basically the variable ${MEMBERINTERFACE} will have the extension (if
 using dynamic members) or the agent number.

 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez Prats
 Director de Tecnología
 +52-55-91169161 ext 2001

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Re: [asterisk-users] Receiving musinc on hold instead of ring

[Sam Govind]

Very strange indeed! post the dialplan lines as well. Seems like a very
normal Dial command execution. Also complete SIP packets for this particular
behaviour can show some insider. Which version of Asterisk you are using?

On Wed, Sep 28, 2011 at 6:44 AM, Alejandro Recarey alexreca...@gmail.comwrote:

 Hi all and thanks for reading.

 I am having a very strange issue. When dialing out with a certain
 carrier, asterisk 1.6.20 will play music on hold instead of a ring
 tone, although this behaviour is NOT what I want.

 Example dialplan execution:

 -- Executing [0021266xxx@test:13] Progress(SIP/100-1e04, ) in new
 stack
 -- Executing [0021266xxx@test:14]
 Dial(SIP/100-1e04,SIP/21266xxx@x.x.x.x) in new stack
 -- Called 21266xxx@x.x.x.x
 -- Call on SIP/x.x.x.x-1e05 placed on hold
 -- Started music on hold, class 'default', on SIP/100-1e04
 -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04

 Now, a SIP packet capture shows no trace of the call being put on hold!

 Sample wireshark capture for the same call:

 x.x.x.x - y.y.y.y SIP/SDP Request: INVITE sip:21266xxx@x.x.x.x, with
 session description
 y.y.y.y - x.x.x.x SIP Status: 100 Giving a try
 y.y.y.y - x.x.x.x SIP/SDP Status: 180 Ringing, with session description

 And I get the music on hold instead of the ringtone. I have tried
 placing Progress() in front of Dial() but to no avail. I do not want
 to use the r option in Dial() because then I lose the destination
 ringtone in early media which is important to my customers.

 Anybody had a similar issue? Any idea of what parameters I can try to
 tweak, as I am stumped...

 Thanks!

 Alex

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Re: [asterisk-users] Call does not pass through

[Sam Govind]

I see a couple of conflicting extensions as well as something I assume
copy-paste malfunction. Please paste the CLI logs of the call.

On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito
mr...@mail.altcladding.com.phwrote:

  Thanks All. Here is my config:

 *On my Firewall NAT:*

 *I allowed the following ports: 4569,5004-5082, 1-2*
 *
 On Asterisk Box:*

 Here is the extensions.conf:
 *[general]
 static=yes
 autofallthrough=yes

 [avaya-internal]
 exten = s,1,Answer()
 exten = s,2,background(pls-entr-num-uwish2-call)
 exten = s,3,WaitExten()
 exten = s,4,Dial(SIP/${EXTEN})
 exten = s,5,Dial(H323/${EXTEN})
 exten = s,6,PlayBack(vm-nobodyavail)
 exten = s,7,HangUp()

 exten = 1000,1,Dial(SIP/1000)
 exten = 1000,1,Answer()

 exten = 1000,2,PlayBack(vm-goodbye)
 exten = 1000,3,HangUp()

 #Extension for recording
 exten = 9000,1,Answer()
 exten = 9000,2,Background(pm-to-record-phrase)
 exten = 9000,3,Hangup()
 #exten = 9000,3,Wait(2)
 exten = 9000,4,Record(alt_recording%d:ulaw)
 exten = 9000,5,Wait(2)
 exten = 9000,6,Playback(${RECORDED_FILE})
 exten = 9000,7,Wait(2)
 exten = 9000,8,Hangup

 exten = _,1,Dial(SIP/${EXTEN}@Avaya)
 exten = _11XX,1,Dial(H323/${EXTEN}@Avaya)

 exten = _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
 exten = _X,2,Dial(SIP/${EXTEN}@cordia)

 exten = _,1,Authenticate(/etc/asterisk/passcode.txt,a)
 exten = _,2,Dial(SIP/${EXTEN}@cordia)*



 Regards,
 Malvin


 On 9/26/2011 9:56 PM, Ruben Rögels wrote:

 Am 26.09.2011 13:18, schrieb Malvin Rito:

  Hi list,
 My call does not pass through on the first dial, I have to redial again
 the number for the call to pass through. I'm not sure if the problem is
 on my voip proovider or my asterisk server setup. Any thoughts pls?

 Regards,
 Malvin

  Hi,

 could be a NAT related issue.

 Please be more specific about your setup.

 best regards,
 Ruben

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Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[Sam Govind]

Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do
tell more about it.

On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 Actually it doesn't say AGI(async:script) it says AGI(async:agi) and
 than continues further to setting up an AMI user so the script is executed
 through the manager interface?? Than it says AGI(agi:async).?? Well most
 importantly it says Cons of async AGI: It is the most complex method of
 using AGI to implement. ..:) I have been interested in Async AGI as well
 and after reading your post looked into the link you provided, seems
 different than what we immediately think, a background process.

 Perhaps just start the script normally AGI(script.sh) and than inside it
 run your background process background-script.sh  /dev/null 21 
 /dev/null  or fork a new process, detach, run in background, etc...

 Hopefully somebody else can point us towards the right direction in setting
 up a real asterisk asynchronous AGI application.

 --
 Mehmet

 On Sep 25, 2011, at 2:00 AM, Randall Degges wrote:

  Hi Everyone,
 
  I've been trying to get asynchronous AGIs working in some Asterisk code I
 have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and
 AGI scripting overall. Here's my problem: I can't get Asterisk to execute
 *any* AGIs asynchronously.
 
  Firstly, I discovered asynchronous AGIs via Asterisk: The Definitive
 Guide. The asynchronous AGI information I read can be found online, here:
 http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the
 section titled Async AGI--AMI Controlled AGI).
 
  According to the book, since Asterisk 1.6.0 the AGI dialplan application
 has been able to execute AGI scripts asynchronously, via the syntax:
 
  exten = s,1,AGI(async:script)
 
  According to the book, using the async: prefix should have Asterisk run
 the AGI script in the background and instantly continue executing dialplan
 code.
 
  So here's my Asterisk dialplan code that's being run:
 
  [hangup]
  exten = s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
  exten = s,n,Return()
 
  Pretty simple context--essentially my AGI script just does some call
 clean up logic before a caller hangs up, talking to a few web servers and
 generating statistics for later usage. What happens when Asterisk executes
 this context, is:
 
  WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute
 '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does
 not exist.
 
  As you can see, Asterisk is ignoring the async: directive, and treating
 it as part of the AGI script path.
 
  Is there anyway for me to make asynchronous AGIs work? I've tried
 searching online to no avail.
 
  I'd greatly appreciate any responses, thanks for your time.
 
  -Randall
 
  --
  Randall Degges
  http://rdegges.com/
 
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Hmmm..interesting..I haven't came across anything like this so far..How
about making a new table for the insertion of a new call data..and trigger
some script to activate AMI/Call file according to new call data.

http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10

On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis sym...@gmail.com wrote:

 Hello Everyone,

 I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
 wondering if it Is possible to have Asterisk make a calls based on a
 record inserted in a table realtime? If I have to develop something using
 AGI
 or AMI, I can do this  with a little direction?

 Thanks in Advance,

 Nick

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Re: [asterisk-users] AGI Problem

[Sam Govind]

How much time your AGI is taking? Check if it is completing its task and not
killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call
channel and related tasks.

On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
  I have an AGI script that occasionally disappears without completing its
 action and asterisk logs the following.
 
   Local/0123456@context-f46e;1AGI Script script.php completed,
 returning 4
   Spawn extension (context, 0123456, 2) exited non-zero on
 'Local/0123456@context-f46e;1'


 I also changed the dialplan and added a line to print AGISTATUS, but when
 this returning 4 happens, asterisk stops there and doesn't execute any
 further dialplan actions, so I don't even see AGISTATUS value.

 exten = h,1,AGI(script.php,${ANSWEREDTIME},,,)
 exten = h,n,NoOp(${AGISTATUS})

 Executing [h@context:1] AGI(Local/0123456@context-4b79;1,
 script.php,13,,,) in new stack
 Local/0123456@context-4b79;1AGI Script script.php completed, returning 4
 Spawn extension (context, h, 1) exited non-zero on
 'Local/0123456@context-4b79;1'

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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

[Sam Govind]

yes you are right. You should put it before calling in the queue. Set
monitor filename as you want. Also you can set directory in the filename
here too. So when the queu is triggered and mixmonitor starts it'll use that
filename and record inot the file.


On Sat, Sep 24, 2011 at 2:08 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi All;

 I noticed in the queues.conf the configuration for recording the calls in
 the queuing, and regarding to the filename (or any other parameter), it is
 written that I can determine the filename using the command:

 Set(MONITOR_FILENAME=foo)


 But it should be called from the dialing plan, but really i did not
 understand how to call it from the dialing plan.

 Well, for example this is my dialing plan to route for the queuing, how I
 can set the filename:

 [CustomerSupport]

 include = Internal

 exten = s,1,Queue(CustomerSupport,t,,,120)
 exten = s,2,Macro(voicemail,SIP/reception)

 By the way, I need in the filename to appear the following:
 The SIP username for the IP Phone that the call is routed for it
 The calling number
 The Time of the call

 Actually for the outbound recording, I am using the below command (I
 mentioned it to declare the time format I am using and to declare how the
 filename to be named):

 exten =
 _9Z,1,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)

 So I hope if someone can help me to write the Set(MONITOR_FILENAME=foo) in
 a way to acheive same format the filename of the recorded outgoing calls (in
 addition that until now I am not able to know where I have to place the
 Set(MONITOR_FILENAME=foo).


 For example, should I place it as following:
 exten = s,1,Set(MONITOR_FILENAME=.)
 exten = s,2,Queue(CustomerSupport,t,,,120)
 exten = s,3,Macro(voicemail,SIP/reception)

 Appreciate if someone help me plz.
 Regards
 Bilal

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Re: [asterisk-users] AGI Problem

[Sam Govind]

Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
asterisk -cvg and then make a call.. see whats your AGI doing in there
!!

On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 Thanks for the response.

 Asterisk logs the execution of the AGI and script completed messages within
 the same second, so less than a second.

 --
 Mehmet

 On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:

 How much time your AGI is taking? Check if it is completing its task and
 not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all
 call channel and related tasks.

 On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
  I have an AGI script that occasionally disappears without completing its
 action and asterisk logs the following.
 
   Local/0123456@context-f46e;1AGI Script script.php completed,
 returning 4
   Spawn extension (context, 0123456, 2) exited non-zero on
 'Local/0123456@context-f46e;1'

 I also changed the dialplan and added a line to print AGISTATUS, but when
 this returning 4 happens, asterisk stops there and doesn't execute any
 further dialplan actions, so I don't even see AGISTATUS value.

 exten = h,1,AGI(script.php,${ANSWEREDTIME},,,)
 exten = h,n,NoOp(${AGISTATUS})

 Executing [h@context:1] AGI(Local/0123456@context-4b79;1,
 script.php,13,,,) in new stack
 Local/0123456@context-4b79;1AGI Script script.php completed, returning
 4
 Spawn extension (context, h, 1) exited non-zero on
 'Local/0123456@context-4b79;1'


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Re: [asterisk-users] AGI Problem

[Sam Govind]

Should we consider it ignorable or you are still interested in resolving !
Best of Luck in any case !

On Sat, Sep 24, 2011 at 6:16 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 Out of about 18000 instances this problem occurred 48 times yesterday. So
 it has been pretty much impossible for me to replicate it under test
 conditions.

 Thanks

 --
 Mehmet

 On Sep 24, 2011, at 4:00 PM, Sam Govind wrote:

 Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
 asterisk -cvg and then make a call.. see whats your AGI doing in there
 !!

 On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 Thanks for the response.

 Asterisk logs the execution of the AGI and script completed messages
 within the same second, so less than a second.

 --
 Mehmet

 On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:

 How much time your AGI is taking? Check if it is completing its task and
 not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all
 call channel and related tasks.

 On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu meh...@activecom.netwrote:


 On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
  I have an AGI script that occasionally disappears without completing
 its action and asterisk logs the following.
 
   Local/0123456@context-f46e;1AGI Script script.php completed,
 returning 4
   Spawn extension (context, 0123456, 2) exited non-zero on
 'Local/0123456@context-f46e;1'

 I also changed the dialplan and added a line to print AGISTATUS, but when
 this returning 4 happens, asterisk stops there and doesn't execute any
 further dialplan actions, so I don't even see AGISTATUS value.

 exten = h,1,AGI(script.php,${ANSWEREDTIME},,,)
 exten = h,n,NoOp(${AGISTATUS})

 Executing [h@context:1] AGI(Local/0123456@context-4b79;1,
 script.php,13,,,) in new stack
 Local/0123456@context-4b79;1AGI Script script.php completed, returning
 4
 Spawn extension (context, h, 1) exited non-zero on
 'Local/0123456@context-4b79;1'


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Re: [asterisk-users] Change of default IVR prompt for meetme conference bridge.

[Sam Govind]

UmmmWhen I was a child I replaced the prompts to do that, Now I'd
suggest you to try creating a new directory in /sounds folder like /en i.e
/meetme and put in corresponding prompts there.
Then just before going into the meetme application change the Language for
the current call in dial plan like this.

exten = le-going-to-meetme,n,SET(CHANNEL(language)=meetme)
same =  n,Meetme(${some-conf-room},${itsoptions})


Try this and let me know the results. Make sure that the actual sound files
are placed on equal directory level as they are currently in
sounds/en/meetme-files.wav or w/e

I hope this be of some help to you.

Thanks and regards,
- Sammy


On Thu, Sep 22, 2011 at 7:21 AM, NaJIm getna...@gmail.com wrote:

 Hi,

 Is it possible to change the default voice prompt for Asterisk meet me
 conference bridge. We have our own customized recordings for Welcome and PIN
 request and would like to use that instead of the default Please enter
 your..  .

 If I replace the default sound file with my custom file by using the
 same filename as the default message, will it affect any other
 applications..??

 Thanks,
 Najim


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Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
if Voicemail PIN are required to be used use application
*MAuthenticate([mailbox][@context][,options]
*

Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org wrote:

 Something like this should work:

 exten = _011.,1,Answer
 exten = _011.,n,Wait(1)
 exten = _011.,n,Read(password,enter-password,5)
 exten = _011.,n,GotoIf($[${password} = 12345]?5:9)

 exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
 exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

 exten = _011.,n,Hangup
 exten = _011.,n,Playback(invalid)
 exten = _011.,n,Hangup

 Could be cleaned up (the GotoIf isn't very descriptive about where it's
 going), but it's a starting point.


 On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:

 Hi List,
 I currently have a asterisk server running used for dialing-out for IDD but
 I want to Put a pincode wherein only users with the right pin code will be
 allowed to dial IDD. Any sample dialplan you can suggest pls?

 Thanks,
 Malvin
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Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

[Sam Govind]

+1 Dale
Alternatively I'd troubles using the MixMonitor() command execution, so what
I did is used System(my commands here) just after the StopMixMonitor().
Using StopMixMonitor() is always recommended to guarantee save the recorded
file and using any commands via System() is easy.

On Wed, Sep 21, 2011 at 6:57 AM, Dale Noll dn...@wi.rr.com wrote:

 **
 I am not real familiar with the size of MixMonitor parameters, but just
 looking at the output, I would suggest you change the logic to call a script
 with a single argument.
 something like this,

 MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME})

 --- /usr/local/bin/convert_to_mp3 --
 #!/bin/bash

 WAV=$1
 MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
 /usr/bin/lame ${WAV} ${MP3} -b 16 -s 9.6 -m m --bitwidth 8 --lowpass
 9.6 --resample 8 --lowpass-width 1  rm -f ${WAV}

 --- end of script ---
 Set the permissions so it is executable by the asterisk owner.

 Note:  This has not been tested and is intended as a starting point.


 Dale



 On 09/20/2011 07:53 PM, Ikka - Mitra Kreasindo wrote:

  Is anyone can help me with this ? I’m really desperate…

 ** **

 Thx in ad.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
 *On Behalf Of *Ikka - Mitra Kreasindo
 *Sent:* Wednesday, September 14, 2011 5:02 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* [asterisk-users] Mixmonitor command parameter problem on
 Asterisk 1.8.4

 ** **

 Dear all…

 ** **

 I’m using MixMonitor command in my dialplan, and I used the “command”
 parameter to execute some thing after recording the file.

 ** **

 I used the command parameter to convert the wav file that created earlier
 to MP3 and than deleted the WAV file.

 ** **

 It worked fine with asterisk 1.4.21.2. and 1.6x

 But than I have a new asterisk server with asterisk 1.8.4. The command
 parameter doesn’t work. It’s trimed for about 297 character only. The rest
 was gone. 

 ** **

 This is part of the log with Asterisk 1.4.21.2

 ** **

   -- Executing [08129981925@speedy:7] MixMonitor(SIP/10001-b7d71bd0,
 /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav|bW(2)|/usr/bin/lame
 /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav
 /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.mp3
 -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
  rm -f
 /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav)
 in new stack

 ** **

 This is part of the log with Asterisk 1.8.4

 ** **

   -- Executing [08129981925@speedy:7] MixMonitor(SIP/10001-001a,
 /var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav,bW(2),/usr/bin/lame
 /var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav
 /var/spool/asterisk) in new stack

 ** **

 ** **

 As you can see, with 1.8.4 the command paramater is trimed… 

 ** **

 Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is there a
 quick workaround for this problem ? 

 ** **


 --
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  Lyta Alexander - Babylon 5


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Re: [asterisk-users] oddity with CISCO CCM and Asterisk

[Sam Govind]

Hi Danny,
If you explain some more about this phantom process !! I've never seen
asterisks doing this before. This initiation of a new call is always
dependent upon arrival of an INVITE. I doubt its CCM that is doing some
re-INVITES or sort of keepalive for this call and thus a phantom call is
created !



On Mon, Sep 19, 2011 at 10:58 PM, Danny Nicholas da...@debsinc.com wrote:

 Hi List,

  I have a system that connects into Asterisk 1.4.41 using CISCO
 CCM 7.  Everything works great except when a call is transferred to the
 operator.  The call goes to the operator via a native bridge and is
 completed, then a “phantom process” starts and tries to launch a new call
 every 15 minutes.  I modified the dialplan to hangup these phantom calls,
 but no still joy.  I get this message:

 [Sep 19 10:32:47] NOTICE[14249] chan_sip.c: Call from 'XXX' to extension
 'X' rejected because extension not found.  

 ** **

 14249 is not showing as running, but it is accessible vi pmap 14249

 ** **

 Could this be a problem with pbx_loopback?

 ** **

 Thanks

 Danny Nicholas

 ** **

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Re: [asterisk-users] Message recorder

[Sam Govind]

Hey Steve,

I very much appreciate your reply, I created that dialplan the same day but
after 8~10 hours of working. Yes, I use combination of MixMonitor and
Monitor.
I'll share my logical dialplan here.

It works like: MixMonitor() is used with option a (append) to record all
user recordings in one file. Monitor is only for most recent recording,
overwrites on each retry. This had to be Gosub for user to press * or #. On
* I jump back to Mixmonitor and monitor and wait for user input at
Waitexten(300,m(silence)). on # just concatenate the required Audio with
previously required audio file.

its something like this (psuedo-code)

[recorder-gosub]
exten = s,1,NOOP(I'm trying to record your voice - Previously recorded
final messages are in file ${GOODAUDIO})
same =  n,SET(ALLAUDIO=bigfilename)
same =  n,SET(THISAUDIO=small-file)
same =  n(rerecord),Mixmonitor(${ALLAUDIO},a)
same =  n,Monitor(${THISAUDIO})
same =  n,Waitexten(300,m(silence))

exten = i,1,GOTO(*,1)
exten = t,1,GOTO(*,1)

exten = *,1,GOTO(s,rerecord) ; user wants to retry recording message

exten = #,1,StopMonitor()   ; user thinks his message is god enough and
save message now
same = n,StopMixmonitor()
same = n,System(sox ${THISAUDIO} ${GOODAUDIO} ${GOODAUDIO})  ;This will
concatenate this required recording to previously recorded good responses.
same = n,Return()

P.S: Macros are useless ! Use Gosub instead. Calling a Gosub inside a macro
(just to get user DTMF) clears macro stack and leaves you look like
stupified.

Thanks,
- Sammy

On Sun, Sep 18, 2011 at 2:13 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Sat, 17 Sep 2011, Sam Govind wrote:

  Requirement: Two copies of the recorded message are required.


  [Recorder-A] One will contain only the last message recorded(final)


  [Recorder-B] second one will record all the previous retries of the
 recording.

 Once the instruction file is played sound recording will start.

 Meanwhile recording if user press * Instruction file is played again and
 message recording: continues for Recorder-B and restarted for Recorder-A.

 If user presses # meanwhile recording..Save both files and continue to
 next extensions.

 I've done sort of combination with Record() application, Mixmonitor,
 Monitor etc but nothing successful so far !


 I don't think you can have a single call recording an 'attempt' while you
 record the entire call at the same time. How about recording each attempt
 separately and concatenating after the fact?

 I did something similar a few years ago using an AGI with the 'record file'
 AGI command and exec'ing the 'monitor' and 'stopmonitor' applications.

 At the completion of the call, an AGI executed 'normalize' to adjust
 different caller's 'volume' on the individual files, 'sox' to concatenate
 the files, 'ffmpeg' to encode to WMA (client requirement) and 'curl' to
 upload to the client's web site.

 FYI, if you anticipate concatenating more than 32 files, you'll need sox
 14.x instead of sox 12.x included with some distributions.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] DTMF problem

[Sam Govind]

Hey there,

I don't think that its DTMF mode issue ! OP say pressing 9 asterisk ignores
while pressing 6 is OK. Using expensive PBX solutions should be never be the
first priority.

So I'd a similar experience in some asterisk version when I used to enter 2
asterisk always took 3-4 seconds to do anything wheras all other DTMF digits
were working as quickly as DTMF entered.
Since pressing 6 key works fine means this could be more possibly issue with
handset.
Also there is an option in background application m where background will
only accept the DTMF whose extens are created in the same context..so if
you've something like this in your dialplan

[test-BKGRND]
exten = s,1,Background(som-sound-file,m)
exten = s,n,Waitexten(20)

exten = 1,1,NOOP(User presssed 1)
exten = 3,1,NOOP(User presssed 1)
exten = 5,1,NOOP(User presssed 1)

Background will act as only recognizing DTMF 1,3,and 5.

even if its DTmfmode issue ...you can one more trick to fix this as well..
use application sipdtmfmode(inband|rfc2833|info)  if call is coming from a
particular caller/UA.

I hope this could be of some help.

Regards,

- Sammy

On Mon, Sep 19, 2011 at 4:51 AM, Zeeshan A Zakaria m...@visionvoip.comwrote:

 This DTMF problem has always been there and there is no real solution for
 it, other than using those expensive PBX systems like that from Avaya,
 Cisco, etc. This problem happens when you are sending inband DTMF tones. Via
 softphone you are sending out-of-band DTMF which is basically SIP messages.

 --
 Zeeshan A Zakaria

 IT Consultant
 www.zeeshanz.com
 855-ZEESHAN

 asterisk asterisk aster...@ck-lee.com wrote:

 From time to time, I have a DTMF problem. The asterisk cannot recognize
 my
 handset key pressed if I press 9 to start with. However, if I press
 with 6,
 it is ok.
 
 On the other hand, if DMTF key is generated from softphone, it is ok.
 
 My dialplan is as follow
 
 exten = 1002,1,Answer
 exten = 1002,n,Wait(2)
 exten = 1002,n,Background(thank-you-for-calling)
 exten = 1002,n,Background(vm-enter-num-to-call)
 exten = 1002,n,WaitExten()
 exten = 1002,n,Hangup
 exten = i,1,Background(pbx-invalid)
 exten = i,2,Goto(1002,1)
 exten = t,1,Background(vm-goodbye)
 exten = t,2,Hangup
 
 Thanks for the help in advance.
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 Sent from my Android phone with K-9 Mail. Please excuse my brevity.

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Re: [asterisk-users] single registration per user

[Sam Govind]

Hmmm..this could be a complex solution - Use OpenSIPS to handle
registration. On each new register attempt see if a user AOR or other
records exists already - if yes deny registration.

On Mon, Sep 19, 2011 at 1:23 AM, Catalin S. jonsonpla...@gmail.com wrote:

 Hello Eric,

 Is about outgoing calls from multiple devices with the same username at
 aprox same time. The overwritten is for incomming calls. I want to prevent
 using the same account in multiple devices at same time. The solution with
 IP will not apply because users may be behind nat or will change everytime
 multiple access points. Do you have any other clues?

 Thank you for answers,
 Best regards.


 On Sun, Sep 18, 2011 at 8:37 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Asterisk only allows one device per peer to register.  If a 2nd device
 registers, the first registration is overwritten.

 You can use permit/deny to limit which IPs a device can register from.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
 Sent: Sunday, September 18, 2011 4:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] single registration per user

 Hello,

 I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock
 every extension to a single registration per device. Many of users tried to
 log on my asterisk from 2, 3 devices and I want allow only one.
 Is there any solution for fix this?

 Thank you.

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Re: [asterisk-users] Sip re-register / delay problem.

[Sam Govind]

Hey,

I don't think there could be any solution to this. Even a SIP Proxy...I
don't think so you'll get enough control there to get re-registers from
lagging users only. SIP Timers adjustment on each user level is something
atleast I haven't cam across so far. SIP Timers are global params for all.

btw,
registerattempts = 5
registertimeout = 5

are used when asterisk is registering out to some other server and not for
asterisk clients.

Regards,
-Sammy

On Sun, Sep 18, 2011 at 1:15 PM, Catalin S. jonsonpla...@gmail.com wrote:

 Hello,

 Can someone help me with some tips on this?

 many thanks


 On Wed, Sep 14, 2011 at 5:03 PM, Catalin S. jonsonpla...@gmail.comwrote:

 Hello,

 For the moment I have the following settings in my sip.conf. I want to
 optimize them to archive the following things:

 - for the moment all my users will re-register too often. I want that only
 lagged users to re-register quickly.
 - check from time to time all users but no too often to see if is logged
 and can be called.

 Overall i want only lagged users to reregister and users with good
 response time to be check from time to time.

 defaultexpiry = 900
 defaultexpirey = 900
 maxexpiry = 300
 maxexpirey = 300
 minexpiry = 60
 registerattempts = 5
 registertimeout = 5
  rtpholdtimeout = 900
 rtptimeout = 60
 jbmaxsize = 60
 jbresyncthreshold = 200
 qualify = yes
 qualify = 600
 qualifyfreq = 60

 Thank you.

 P.S. If you consider that i use too much options you can tell me what to
 drop. I use asterisk 1.8.6.0.



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[asterisk-users] Message recorder

[Sam Govind]

Hello List,

Good day. I'm trying to create a message recorder. User will be prompted
with some soundfile (instructions on how to use recorder * to restart and #
to submit).

*Requirement:* Two copies of the recorded message are required.
[Recorder-A] One will contain only the last message recorded(final)
[Recorder-B] second one will record all the previous retries of the
recording.

Once the instruction file is played sound recording will start.

Meanwhile recording if user press * Instruction file is played again and
message recording:  continues for Recorder-B and restarted for Recorder-A.

If user presses # meanwhile recording..Save both files and continue to next
extensions.

I've done sort of combination with Record() application, Mixmonitor, Monitor
etc but nothing successful so far !

Please help.

Thanks.

- Sammy
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Re: [asterisk-users] testing simultaneous calls

[Sam Govind]

A little  look at the dialplan which rings your extension, or get dtmf, and
plays DTMF will help better understand. btw you can set the
context/extension/priority in a call file to skip some priorities of a
particular extension set.

On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON e...@lanline.com wrote:

 ** **

 Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live
 production.

 ** **

 I cannot run valgrind and I do not have the right flags set in menuselect.
 

 ** **

 I can however at the dead of the night run stress tests.

 ** **

 I want to simulate x-amount of concurrent calls to both a dtmf dialplan,
 which is working, as well as MoH dialplan to see if this could be the cause
 of crashing.

 ** **

 How do I test this?

 Is it a call file that can handle this without ringing my extension first,
 like internal system calling?

 ** **

 Thanks,

 --Eric

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Re: [asterisk-users] Monitoring second leg being dialed?

[Sam Govind]

The image you provided didn't open so I'm not sure about the design. If you
can send some SIP flow diagram and Asterisk CLI logs maybe it'll help
understand the problem.

On Fri, Sep 16, 2011 at 1:28 AM, Gilles codecompl...@free.fr wrote:

 Hello

My ISP provides an FXS port to plug a handset, which can be used to
 make free calls to (GSM) cellphones, similar to the Billion ADSL
 modems:

 http://au.billion.com/product/voip.php

 My plan is to install an SIP client on a smartphone, so that when I'm
 travelling, I can connect to a good wifi hotspot, register with an
 Asterisk server at home which has an FXO card, tell Asterisk the
 number I wish to dial, and have it dial out through the FXO card and
 the FXS port on the ADSL modem.

 Here's the diagram:

 http://img844.imageshack.us/img844/3308/asterisksippstncallback.png

 Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1)
 when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the
 call answered although there's no actual phone connection yet, and
 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was
 answered (and if yes, by a live human being rather than an answering
 machine) or if the line is still ringing.

 A so-so solution is to simply tell Asterisk to loop through a voice
 message (This is a call from Joe Allen. Please hit any key and you
 will be connected), so we know that a human being has answered the
 call, but I was wondering if there were a better solution.

 Is it possible for Asterisk to somehow play on channel #1 what's
 happening on channel #2 while Dahdi/Zaptel is actually still dialing,
 so that I handle call progression manually from my cellphone and the
 callee doesn't end up hearing that odd recorded message?

 Thank you.


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Re: [asterisk-users] Inter-astersik dialling encounteres no audio

[Sam Govind]

This obviously is pointing to NAT issue. see if you've configured both
asterisk servers with externip= PUBLICIPOFAsterisks.

Studying SIP traces on each console and specially looking at the SDPs in
INVITE will help you find out exact problem. I expect that one of the
asterisk box is sending the audio to its LAN/Private IP whereas it should be
sending RTPs to Public IP of other Asterisk.


On Fri, Sep 16, 2011 at 12:50 PM, Lee, John (Sydney) john@compuware.com
 wrote:

 **

 I have been deploying Asterisk (open source PABX) in the company which I
 work.

 So far, all the Asterisk servers do not really talk to each other.
 Recently, I am experimenting to dial from one Asterisk server to another
 through the WAN and I encountered a no-audio problem although the callee's
 phone can ring.

 I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
 allowed to go through but not RTP (UDP 16384-32767).



 Case A

 ==

 This is a simplified diagram of how I am testing the dialling between 2
 subnets.

 In this case, phone A is registered in Asterisk A and phone B is
 registered in Asterisk B.

 Phone A -- Asterisk A -- Router A == WAN == Router B --
 Asterisk B -- Phone B



 Case B

 ==

 However, before I have tested successfully using this kind of connection.

 In this case, phone B1 and B2 are registered in Asterisk B although they
 are on different subnets.

 Both phone B1 and B2 can ring and audio is allowed to pass through.

 Phone B1 -- Router A == WAN == Router B -- Asterisk B --
 Phone B2



 I am mystified why audio is allowed go through in case B but not case A.



 Can someone be kind enough to help me to understand why I have this
 problem?

 If the router is blocking RTP traffic, then why is that I have no audio
 problem in case B?

 Thanks in advance.


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Re: [asterisk-users] Monitoring second leg being dialed?

[Sam Govind]

The image just don't open for me, a wild from appears and tells me Domain
blocked bla bla. Try attaching image in this mail.


 Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at
 home connected to their ADSL modem so that they can make free calls
 from overseas?


LOL- Its like asking an army Have you guys ever worked with guns!!  :P

Please try producing SIP traces so your problem could be identified. Which
asterisk and DAHDI version you are using btw?

On Fri, Sep 16, 2011 at 2:51 PM, Gilles codecompl...@free.fr wrote:

 On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com
 wrote:
 The image you provided didn't open so I'm not sure about the design.

 Sorry about that. It's a PNG file and it opens in the two browsers I
 tried.

 The reason I don't simply get a subscription with a VoIP provider and
 must go through an Asterisk server + connection to the FXS port is
 that outgoing calls are free, which is nice when calling cellphones,
 especially when travelling.

   If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll
 help
 understand the problem.

 I haven't done it yet, so have no logs to show.

 I'd simply like to hear what's going on channel #2 while Dahdi is
 still dialing, instead of simply being kept waiting.




 Thank you.


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[asterisk-users] Temporarily disable DTMF on a call

[Sam Govind]

Hello List,

I need help on disabling DTMF from a caller for a specific set of dialplan
commands and enable DTMF for some other dialplan part. This is not a SIP
peer - just live incoming call on SIP.

Please help.

Thanks

-Sammy
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Re: [asterisk-users] Temporarily disable DTMF on a call

[Sam Govind]

Hey :)
Smiles on your reply but Its complicated :P
Anyways  I was actually using SayUNixTime() application and found out that
if a digit is pressed it breaks and go to that extension. So I wanted user
to listen to the time but key presses don't do any harm as well.
I've successfully done it now using application SIPdtmfmode(info) whereas it
was rfc2833 just before hitting the sayunixtime() and then after that reset
sipdtmfmode(rf2833)

Like this.

...
exten = s,n,SIPDtmfMode(info)
exten = s,n,SayUnixTime(${params})
exten = s,n,SIPDtmfMode(rfc2833)
...

Now your x-girlfriend will never know if its her phone's fault or you've
done some trick :P

On Fri, Sep 16, 2011 at 11:39 PM, Danny Nicholas da...@debsinc.com wrote:

 I would do this with ex-girlfriend logic

 ** **

 [mycontext]

 Exten = s,1,playback(instructions)

 Exten = s/5551212,n,goto(end)

 Exten = s,n,read(var,prompt, …)

 Exten = s,n,process..

 Exten =s(end),n,hangup

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sam Govind
 *Sent:* Friday, September 16, 2011 1:03 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Temporarily disable DTMF on a call

 ** **

 Hello List,

 ** **

 I need help on disabling DTMF from a caller for a specific set of dialplan
 commands and enable DTMF for some other dialplan part. This is not a SIP
 peer - just live incoming call on SIP. 

 ** **

 Please help.

 ** **

 Thanks

 ** **

 -Sammy

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Re: [asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call

[Sam Govind]

Hey,

The callee server is complaining too loud Call from '2765' to extension '*
1166:password*' rejected because *extension not found*.
Try changing the Dial string as DIAL(SIP/asterisk-callee/${EXTEN}) or w/e
extension you require in place of ${EXTEN}
Let me know what changes.

Also this is a good read:
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/

Wed, Sep 14, 2011 at 12:37 PM, Lee, John (Sydney) john@compuware.comwrote:

 I was trying to do a SIP call between two Asterisk servers (1.4.21.2)

 ** **

 1) On the caller server, I coded the following in extensions.conf

 Dial(SIP/1166:password@asterisk-callee);

 ** **

 2) On the callee server, I coded the following in sip.conf

 [1166]

 type=friend; Friends place calls and receive calls

 context=incoming   ; Context for incoming calls from this user
 

 host=dynamic   ; This peer register with us

 dtmfmode=rfc2833   ; Choices are inband, rfc2833, or info

 qualify=yes; Monitor latency between Asterisk server
 and phone

 call-limit=99

 username=1166  ; Username to use in INVITE until peer
 registers

 secret=password; Normally you do NOT need to set this
 parameter

 mailbox=1166@default   ; mailbox 5100 in voicemail context
 .default.

 callgroup=1

 pickupgroup=1

 ** **

 The call was unsuccessful as follows.

 

 1) On the caller machine, this is what we got from the console

 -- Executing [1166@incoming:1] Dial(SIP/1166-09d81668,
 SIP/1166:password@asterisk-callee) in new stack

 -- Called 1166:password@asterisk-callee

 -- SIP/asterisk-callee is circuit-busy

   == Everyone is busy/congested at this time (1:0/1/0)

 ** **

 2) On the callee machine, this is what we got from the console,

 [Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite:
 Call from '2765' to extension '1166:password' rejected because extension not
 found.

 ** **

 However, I found out that if I remove “secret=..” from the SIP entry and
 call without the password, then I will be able to call.

 ** **

 Any thoughts?

 ** **

 ** **

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Re: [asterisk-users] Mysql dialplan statement not executed

[Sam Govind]

I expect that your same query when executed directly on MySQL console
executes successfully ! Normally errors in DB queries are printed on CLI but
apparently there is nothing wrong.

On Wed, Sep 14, 2011 at 5:51 PM, Jonas Kellens jonas.kell...@telenet.bewrote:

 **
 Hello,

 I do the following in a macro in the dialplan :

 exten = s,n,MYSQL(Connect connid localhost user password AsteriskDB)
 exten = s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1
 WHERE routeID=${ARG1} AND nr=1)
 exten = s,n,MYSQL(Disconnect ${connid})

 But nothing changes in my database...

 This is the CLI :

 [Sep 14 16:41:04] -- Executing [s@macro-bal:15]
 MYSQL(SIP/vc5-000b, Connect connid localhost user password
 AsteriskDB) in new stack
 [Sep 14 16:41:04] -- Executing [s@macro-bal:16]
 MYSQL(SIP/vc5-000b, Query resultid 15 UPDATE custDB SET active=1
 WHERE routeID=195 AND nr=1) in new stack
 [Sep 14 16:41:04] -- Executing [s@macro-bal:17]
 MYSQL(SIP/vc5-000b, Disconnect 15) in new stack

 Seems OK, no warnings. But the update has not taken place.


 Kind regards,
 Jonas.

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Re: [asterisk-users] broadcast

[Sam Govind]

Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
linkhttp://www.voip-info.org/wiki/view/Asterisk+auto-dial+out).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s@Konference*

[Konference]
exten = s,1,ANSWER()
exten = s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten = s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten = s,n(message),Konference(43689956,ADMRSTV)
exten = s,n,Hangup()

Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.

Next thing in call file is the destination which will be playing broadcast
message once Konference is called.

*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten = s,1,Answer()
exten = s,n,Set(p=/var/spool/asterisk/monitor/)
exten = s,n,playback(${p}/LQA/12/Biology/Que3)
exten = s,n,playback(${p}/LQA/12/Biology/Que4)
exten = s,n,playback(${p}/LQA/12/Biology/Que5)
exten = s,n,playback(${p}/LQA/12/Biology/Que6)
exten = s,n,playback(${p}/LQA/12/Biology/Que7)
exten = s,n,Wait(10)
exten = s,n,Hangup()

This should work and konference should listen to the playbacks.

Regards,
Sammy.

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.com wrote:

 Hi List,

 I make a script for .call file and then I started playback on local channel
 but nothing was hearing at another channles.

 exten = 1234,1,Answer()
 exten = 1234,n,System(echo -e Channel: Channel: 
 local/23@contest-call\\nContext:
 contest-call\\nExtension: 23\\nPriority: 1  /tmp/${UNIQUEID}.call)
 exten = 1234,n,Konference(43689956,ADMRSTVL)

 [contest-call]

 exten = _X!,1,Answer()
 exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
 exten = _X!,n,Konference(43689956,ADMRSTV)
 exten = _X!,n,Wait(10)
 exten = _X!,n,Hangup()

 in it I am dialing 1234 from softphone then join to conf in mute mode,
 after it .call file start playback at it's own channels but I am not able to
 hear anything into conf.

 As i know localdial is not joining into the conf. but how I will do it so
 that I will be able to hear any played file into conference ?



 On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:

 Good to know,

 I think it'll be a feedback score or a poll from members of the
 conference. So if you use the R option and collect DTMF from members, and an
 AMI script listening to that particular DTMF event collects all. This way
 your AMI listener script should be able to tell you at the end of poll what
 user inserted with DTMF.

 So overall insertion of a broadcast message using Ahmed's method of .call
 file and later on collecting DTMF events from AMI script
 should theoretically work for you.

 On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:

 Hi Sam,

 You are right. I am looking for the same

 On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:

 IMHO, I think Bhaati is trying to get feedback from
 multiple conference users. See DTMF options in Konference module.
  'R' : enable DTMF relay: DTMF tones generate a manager event
  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
 members in the conference

 While some file is played and users press any DTMF collect the AMI
 events from each user and use them as you require.

 Ref:
 http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173


 On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.comwrote:

 Hi Ahmed,

 Konference is also an conferencing application.

 On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
 gohar.ah...@vopium.comwrote:

 Hhhmmm..I dunt have any experience with module Konference. Maybe
 anyone else can help you on that. 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
 bhati
 *Sent:* Monday, September 12, 2011 1:28 PM

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

  ** **

 Hi Ahmed,

 I did the same thing earlier to test the load of Digium card. But this
 time I want to play file and want to get some DTMF from all the members 
 of
 conference.

 So in this case I need more control into Konference module. But when I
 use .call files then control will not go longer with all events.

 Is there any alternate way to do it?

 I appreciate your

Re: [asterisk-users] broadcast

[Sam Govind]

I don't know why you are running into problems.
Once a call file is executed it creates two legs (according to call file
structure) A leg is Channel: Local/1234@conference and once it Answers the
call file the second leg is bridged which should be
Context-Extension-priority. So what I'm asking is make your conference
A-leg and your Playback/messages dial plan B-leg.

take a look at the changes I made to your dial-plan

[conference]
exten = 1234,1,Answer()
exten = 1234,n,Gotoif($[${FIRST-CALLER}  1]?startmsg:pass)
exten = 1234,n(startmsg),System(echo -e
Channel:local/1234@conference\\nContext: contest-call\\nExtension:
23\\nPriority: 1  /tmp/${UNIQUEID}.call)
exten = 1234,n,system(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing)
exten = 1234,n(pass),Konference(43689956,ADMRSTVL)
exten = 1234,n,Hangup()

[contest-call]
exten = 23,1,Answer()
exten = 23,n,Set(p=/var/spool/asterisk/monitor/)
exten = 23,n,playback(${p}/LQA/12/Biology/Que3)
exten = 23,n,playback(${p}/LQA/12/Biology/Que4)
exten = 23,n,playback(${p}/LQA/12/Biology/Que5)
exten = 23,n,playback(${p}/LQA/12/Biology/Que6)
exten = 23,n,playback(${p}/LQA/12/Biology/Que7)
exten = 23,n,Wait(10)
exten = 23,n,Hangup()

Here, changed your script to what I'm thinking. use the above tweak
accordingly. make sure to find out FIRST-CALLER so your tapes start playing
into conference just for once.

-Sammy

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.com wrote:

 Hi List,

 I make a script for .call file and then I started playback on local channel
 but nothing was hearing at another channles.

 exten = 1234,1,Answer()
 exten = 1234,n,System(echo -e Channel: Channel: 
 local/23@contest-call\\nContext:
 contest-call\\nExtension: 23\\nPriority: 1  /tmp/${UNIQUEID}.call)
 exten = 1234,n,Konference(43689956,ADMRSTVL)

 [contest-call]

 exten = _X!,1,Answer()
 exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
 exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
 exten = _X!,n,Konference(43689956,ADMRSTV)
 exten = _X!,n,Wait(10)
 exten = _X!,n,Hangup()

 in it I am dialing 1234 from softphone then join to conf in mute mode,
 after it .call file start playback at it's own channels but I am not able to
 hear anything into conf.

 As i know localdial is not joining into the conf. but how I will do it so
 that I will be able to hear any played file into conference ?



 On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:

 Good to know,

 I think it'll be a feedback score or a poll from members of the
 conference. So if you use the R option and collect DTMF from members, and an
 AMI script listening to that particular DTMF event collects all. This way
 your AMI listener script should be able to tell you at the end of poll what
 user inserted with DTMF.

 So overall insertion of a broadcast message using Ahmed's method of .call
 file and later on collecting DTMF events from AMI script
 should theoretically work for you.

 On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:

 Hi Sam,

 You are right. I am looking for the same

 On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:

 IMHO, I think Bhaati is trying to get feedback from
 multiple conference users. See DTMF options in Konference module.
  'R' : enable DTMF relay: DTMF tones generate a manager event
  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
 members in the conference

 While some file is played and users press any DTMF collect the AMI
 events from each user and use them as you require.

 Ref:
 http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173


 On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.comwrote:

 Hi Ahmed,

 Konference is also an conferencing application.

 On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
 gohar.ah...@vopium.comwrote:

 Hhhmmm..I dunt have any experience with module Konference. Maybe
 anyone else can help you on that. 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
 bhati
 *Sent:* Monday, September 12, 2011 1:28 PM

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

  ** **

 Hi Ahmed,

 I did the same thing earlier to test the load of Digium card. But this
 time I want to play file and want to get some DTMF from all the members 
 of
 conference.

 So in this case I need more control into Konference module. But when I
 use .call files then control will not go longer with all events.

 Is there any alternate way to do it?

 I appreciate your suggestion and will doing in parallel at higher
 priority

 On Mon, Sep 12

Re: [asterisk-users] Asterisk Manager Interface (AMI)

[Sam Govind]

Hey,

I think I remember the same post before. previously I heard someone telling
to use vicidial or some other thing  like that.But I don't think that those
are totally AMI based call-generators.

What I'd recently done is make a php page which connects to Asterisk's AMI
port. I send page request with destination number as parameter and depending
upon the HTTP arguments it send an ORIGINATE event to Asterisk with the
destination number to be dialled out via DAHDI(PRI) and once the call is
answered bridge it to a local dial plan extension which in term played a
sound-file/message to the connecting number.

So whenever I want Asterisk to initiate a call I send a HTTP request to my
Web-Server(hosting Asterisk) a call originated and played a message. You can
choose your design and directly connect to AMI and keep on sending ORIGINATE
events until you've all 200 channels occupied.

Hope it will help.

On Tue, Sep 13, 2011 at 6:26 AM, Kaushal Shriyan
kaushalshri...@gmail.comwrote:

 Hi,

 I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels
 (25 channels per PRI). Can someone please help me understand using
 Asterisk Manager Interface (AMI) available in Asterisk to dial out 200
 numbers and run a campaign for 200 numbers concurrently and play a mp3
 file ?

 Please suggest/guide.

 Regards,

 Kaushal

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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Sam Govind]

1- *-bash: obd-demo.ulaw: No such file or directory* // try use absolute
file path i.e /usr/src/mymp3.mp3 . I guess that's why its saying no such
file or directory.
2-  http://lists.digium.com/pipermail/asterisk-users/2006-March/144689.html Go
through this thread.
3-  When everything fails from sox - libraries dependencies issues  I use
http://www.nch.com.au/switch/index.html this converter. This can help you
for some time for free.

On Tue, Sep 13, 2011 at 5:12 AM, Kaushal Shriyan
kaushalshri...@gmail.comwrote:

 Hi,

 Can someone please comment about the below issue

 [root@host0040 kaushal]# file obd-demo.mp3
 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural
 [root@host0040 kaushal]# sox obd-demo.mp3 -e stat
 sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

 [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w
 vm-intro.ulaw
 sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

 [root@host0040 kaushal]# sox -v 0.125 -V obd-demo.mp3 -t au -r 8000
 -U -b -c 1 obd-demo.ulaw resample -ql
 -bash: obd-demo.ulaw: No such file or directory
 [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1
 obd-demo.ulaw resample -ql
 sox: Failed reading obd-demo.mp3: Do not understand foReply
 rmat type: mp3

 [root@host0040 kaushal]#

 When i invoke the same obd-demo.mp3 it works perfectly fine

 host0040*CLI channel originate DAHDI/g0/xx Application
 MP3Player /home/kaushal/obd-demo.mp3
 [Sep  9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request:
 sig_pri_request 1
 [Sep  9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME:
  NUM:
  -- Requested transfer capability: 0x00 - SPEECH
  -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on
 DAHDI/i1/9833754756-1

 [root@host0040 ~]# rpm -qa | grep sox
 sox-12.18.1-1.el5_5.1
 [root@host0040 ~]# rpm -qa | grep lame
 lame-3.98.4-1.el5.rf
 lame-devel-3.98.4-1.el5.rf
 [root@host0040 ~]#


 MP3 support in  SoX  is  optional
and requires access to either or both the external
 libmad and libmp3lame libraries.  To see if there is support for Mp3
 run sox -h and
look for it under the list of supported file formats as
 mp3.

 [root@host0040 ~]# sox -h
 sox: Version 12.18.1

 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ]

 gopts: -e -h -p -q -S -V

 fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x

 effect: avg band bandpass bandreject chorus compand copy dcshift
 deemph earwax echo echos fade filter flanger highp highpass lowp
 lowpass mask mcompand noiseprof noisered pan phaser pick pitch
 polyphase rate repeat resample reverb reverse silence speed stat
 stretch swap synth trim vibro vol

 effopts: depends on effect

 Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm
 hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw
 ub ul uw voc vorbis vox wav wve

 Which package contains libmad and libmp3lame libraries available on CentOS
 5.6

 Regards,

 Kaushal

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Re: [asterisk-users] Asterisk is keep on sending Register request

[Sam Govind]

Hey krishnan,

Everything happens for a reason. The most intuitive cause of this issue
seems to be network change. Can you confirm that no change in networking
happened! because your server is sending register requests but not getting
responses. Meanwhile the same server replying to scenarios2 can imply either
at your or the server end is blocking. There could be NAT issue if its not
firewall.

1- Make sure you can ping from your asterisk server to Registrar server. Do
a traceroute as well.
2- Check for any firewalls in between (could be fail2ban/ iptables)
3- Verify that no network changes occured.
4- Call your service provider and tell them that their server is not talking
to your server any more. :P

best of luck.

-Sammy

On Mon, Sep 12, 2011 at 11:06 PM, Gopal krishnan 
gopalakrishnan...@gmail.com wrote:

 Hi,

 *Scenario 1*
 I am trying to register a VoIP trunk, which is keep on sending the register
 request and I am not getting any response from the SIP Server, this I am
 trying from one network.

 *Scenario 2*
 From another network if I try the same VoIP trunk, the account got
 registered.

 One thing here to notice is the same account has already been worked in 
 *Scenario
 1* and now which is not working without any reason.

 Any comments would be much appreciated.

 Regards

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Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

[Sam Govind]

Thats goood ! :) thanks for updating.

On Fri, Sep 9, 2011 at 2:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:

 Hi

 Can you please provide an excerpt of your logs when this happens?

 Regards

 Ish

 On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote:
  Hi sam,
 
  Have solved the problem with your advice. Call drop in 10 seconds without
 disconnecting a-party call. Thank you very much.
 
  [TB]
 
  exten =_X.,1,Wait(${INCOMING_WAIT})
 
  exten =_X.,2,Verbose(TB)
 
  exten =_X.,3,Answer()
 
  exten =_X.,4,Set(mainLoop=0)
 
  ;exten =_X.,5,Set(TIMEOUT(absolute)=5)
 
  exten =_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
 
  exten = _X.,6,Dial(DAHDI/7/
 
  09501032209,100,L(3[:1][:3000])g)
 
  exten =_X.,7,Noop(${DIALEDTIME})
 
  exten =_X.,8,Goto(TB,_X.,1)
 
  exten =_X.,n,Hangup()
 
  Cheers
  Vinod Dharashive
  Sent from BlackBerry® on Airtel
 
  -Original Message-
  From: Sam Govind govoi...@gmail.com
  Sender: asterisk-users-boun...@lists.digium.com
  Date: Wed, 7 Sep 2011 11:53:33
  To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
  Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] (no subject)
 
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 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd

 Office:   0161 660 3062


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Re: [asterisk-users] (no subject)

[Sam Govind]

See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.

here's an excerpt from somewhere:

 ; limit calls to ex-girlfriend to 300 seconds
exten = 123,1,Set(TIMEOUT(absolute)=300)
exten = 123,2,Dial(${EX-GIRLFRIEND})
exten = T,1,Playback(im-sorry)
exten = T,2,Playback(vm-goodbye)
exten = T,3,Hangup(  )


Also see if Dial() command options L(x:y:z), or S(x) work out for you when
combined with option g.

On Wed, Sep 7, 2011 at 7:42 AM, Vinod Dharashive vdharash...@gmail.comwrote:

 Hi team,

 I am trying to find solution to hangup b-party call after 1 min with out
 disconnecting the call of a-party but following dial plan which is
 disconnect both the calls.


 Please suggest me the solution.

 [TB]



 exten = _X.,1,Wait(${INCOMING_WAIT})

 exten =_X.,2,Verbose(TB)

 exten =_X.,3,Answer()

 exten = _X.,4,Set(mainLoop=0)

 exten = _X.,5,Set(TIMEOUT(absolute)=10); set time in  milliseconds

 exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)

 exten = _X.,7,Dial(DAHDI/7/

 09501032209,10,S(60))



 exten = _X.,8,Noop(${DIALEDTIME})

 exten =_X.,9,Goto(TB,_X.,1)

 exten =_X.,n,Hangup()

 Thanks
 Vinod Dharashive
 Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Queue agent login notification

[Sam Govind]

you definitely need to create the file extconfig - take sample from
internet. the DB tables need to be created on your own, take help from
internet pages.

On Wed, Sep 7, 2011 at 6:19 PM, Michael voip.quest...@gmail.com wrote:

 I couldn't find the extconfig.conf file in /etc/asterisk and queue_log
 doesn't exist either (as a file or as a db table). We're using AsteriskNOW,
 so maybe these files/tables were not created.

 How should we add them?

 Thanks.

 On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind govoi...@gmail.com wrote:

 See this link:
 http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

 You'll find similar pages where you can setup to store queue
 logs/events(as Alex mentioned) in MySQL DB and further do your triggers or
 functions on them.


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Re: [asterisk-users] Beginner Question: Remote access

[Sam Govind]

There could be as easy solutions as using teamviewer or use tools like
Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs
static public IPs for the sake of completing the route.

On Wed, Sep 7, 2011 at 5:30 AM, A Dunor alsta...@gmail.com wrote:

 Thanks for the speedy pointer Danny.



 On 9/6/2011 8:27 PM, Danny Nicholas wrote:

 Google for IP-tunneling.

 -Original Message-
 From: 
 asterisk-users-bounces@lists.**digium.comasterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-**boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
 On Behalf Of A Dunor
 Sent: Tuesday, September 06, 2011 7:17 PM
 To: asterisk-users@lists.digium.**com asterisk-users@lists.digium.com
 Subject: [asterisk-users] Beginner Question: Remote access

 Hello list, I am a beginner at asterisk. I want to access my asterisk box
 from my laptop, on a different network (mobile hotspot). The asterisk box
 doesn't have a static ip, how do I connect with it using ssh or other such
 programs?

 Thanks for your guidance guys. It is highly appreciated.

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Re: [asterisk-users] Queue agent login notification

[Sam Govind]

See this link:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

You'll find similar pages where you can setup to store queue logs/events(as
Alex mentioned) in MySQL DB and further do your triggers or functions on
them.


On Wed, Sep 7, 2011 at 10:46 AM, Michael voip.quest...@gmail.com wrote:

 We're using FreePBX and I'm wondering if it's possible to add to the
 login/logout macros a command to execute an AGI/Command to launch an
 external process for that.

 Thanks.

 On Fri, Aug 12, 2011 at 2:30 PM, Alex Vishnev alex9...@gmail.com wrote:

 you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from
 that point on, you can store them or take any other action.
 the other way is to use AMI an monitor for Agent login/logoff events
 On Aug 12, 2011, at 7:06 AM, Michael wrote:

  Hello,
 
  Is there a way to either store login/logout agent information in a
 database or at least send an email when an agent logs in or out of a queue?
 
  Thanks,
 
  Michael


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Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

1- Per my experience I've used DB with configuration files and I was amazed
that Asterisk was taking a union of DB + conf file configurations and
accepting both.So if you just make a simple script or DB function to do file
operation on some event/cronjob you'll be saved.

Moreover, if that still may induce duplication into configurations then DB
replication and redundancy is the best way to cater your failure case. There
are hundreds of how-tos on DB redundancy and failure etc.

2- If you've to move forward with this approach I'll suggest you to read
only part of configuration file corresponding to one user i.e [user-1-area]
and over-write that part only. If a new user then just append. This way file
data loss will be minimized(may even avoided totally).

Those were all my suggestions, if anyone else can add valuable comments to
this.

-
sammy


On Mon, Sep 5, 2011 at 11:45 AM, virendra bhati virbh...@gmail.com wrote:

 Hi Sammy,

 Ans of 1st question:-

 As per my experiance Asterisk realtime(DB) based data will lost when your
 server is creash and you may not take backup of your server's DB.
 If any one know  then plese guide me so that I will start working on it.

 Ans of 2nd question:-

 Your question is correct if more then one user will access these
 configuration files then might be some problem will come.

 For this issue I am just make a connection with server then close it after
 finishing the job. So problem will be avoided 



 On Mon, Sep 5, 2011 at 10:47 AM, Sam Govind govoi...@gmail.com wrote:

 Though this might have been resolved/accomplished already but I've couple
 of questions for Virendra Bhati.

 1- If you are doing this to make new accounts for new users, why couldn't
 you use Asterisk realtime(DB) based configurations of
 Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
 and lots of filing !?

 2- Since its a web-based Filing operations and if multiple users are to
 use the same page for appending/overwriting their configurations wouldn't it
 lead to information being lost when multiple users applying their changes ?
 I wonder how do you handle that ?

 I'm sure I'd more questions when I started writing this response mail but
 now I've forgotten those :P

 Thanks,
 Sammy.

 On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen 
 tzafrir.co...@xorcom.comwrote:

 On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
  Hi list,
 
  I want ot do basic work (add-edit-delete) into asterisk configuration
 files,
  like sip.conf, manager.conf,musiconhold.conf etc.
 
  Please guide me how to configure all these files from from AMI
 connection. I
  am able to login into AMI from Login action but I want to do more task
 in to
  it.
 
  *AMI login:- *
 
  *login.php*
 
  ?php
  $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
  fputs($socket, Action: Login\r\n);
  fputs($socket, UserName: root\r\n);
  fputs($socket, Secret: energy\r\n\r\n);
  ?
  *AMI command:-*
 
  Below commands are for musiconhold.conf. I want to add new MOH context
 into
  it.
  ?php
  include(login.php);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, Filename: musiconhold.conf\r\n);
fputs($socket, Srcfilename: musiconhold.conf\r\n);
fputs($socket, Dstfilename: musiconhold.conf\r\n);
fputs($socket, Action-00: newcat\r\n);
fputs($socket, Cat-00: bhavik\r\n);
fputs($socket, mode: files\r\n);
fputs($socket, directory: /var/lib/asterisk/moh\r\n);
fputs($socket, Reload: yes\r\n);
fputs($socket, ActionID: 9873497149817\r\n);
fputs($socket, Action: Logoff\r\n\r\n);

 You're not really editing. You're writing.

 Note the following:

 * It requires Asterisk to be running, and accessible through the manager
  interface.
 * asterisk.conf may be in a path that is not the configuration
  directory. I'm not sure if this special case is handled.
 * #include are basically handled, but mostly for reading. IIRC the write
  is back to a single file. No idea about #exec, which will probably
  have odd interactions with UpdateConfig. Configuration templates
  ('[section](template)') are also not handled gracefully.

 
  After doing all no success :((

 This is a report of the the thing that did not happen. Next time you ask
 a question, please report what actually does happen (I got the following
 response: ...).

 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

 --
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

Are you talking about AstDB or MySQL as DB backend for asterisk?

On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati virbh...@gmail.com wrote:

 Hi Sammy,

 Thanks for share your experance and provide a new way of Asterisk
 communication with DB.

 Actually I am using this feature only for MOH feature of asterisk right
 now. But I will used it to all the configuration files too as per the needs.


 I am not too much aware abut the Asterisk DB handling that's why I used
 ODBC and mysql connection with asterisk.

 One more thing please tell me how to take backup of Asterisk DB ? It is my
 1st and last things which hurt me.



 On Mon, Sep 5, 2011 at 12:48 PM, Sam Govind govoi...@gmail.com wrote:

 1- Per my experience I've used DB with configuration files and I was
 amazed that Asterisk was taking a union of DB + conf file configurations and
 accepting both.So if you just make a simple script or DB function to do file
 operation on some event/cronjob you'll be saved.

 Moreover, if that still may induce duplication into configurations then DB
 replication and redundancy is the best way to cater your failure case. There
 are hundreds of how-tos on DB redundancy and failure etc.

 2- If you've to move forward with this approach I'll suggest you to read
 only part of configuration file corresponding to one user i.e [user-1-area]
 and over-write that part only. If a new user then just append. This way file
 data loss will be minimized(may even avoided totally).

 Those were all my suggestions, if anyone else can add valuable comments to
 this.

 -
 sammy


 On Mon, Sep 5, 2011 at 11:45 AM, virendra bhati virbh...@gmail.comwrote:

 Hi Sammy,

 Ans of 1st question:-

 As per my experiance Asterisk realtime(DB) based data will lost when
 your server is creash and you may not take backup of your server's DB.
 If any one know  then plese guide me so that I will start working on it.

 Ans of 2nd question:-

 Your question is correct if more then one user will access these
 configuration files then might be some problem will come.

 For this issue I am just make a connection with server then close it
 after finishing the job. So problem will be avoided 



 On Mon, Sep 5, 2011 at 10:47 AM, Sam Govind govoi...@gmail.com wrote:

 Though this might have been resolved/accomplished already but I've
 couple of questions for Virendra Bhati.

 1- If you are doing this to make new accounts for new users, why
 couldn't you use Asterisk realtime(DB) based configurations of
 Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
 and lots of filing !?

 2- Since its a web-based Filing operations and if multiple users are to
 use the same page for appending/overwriting their configurations wouldn't 
 it
 lead to information being lost when multiple users applying their changes ?
 I wonder how do you handle that ?

 I'm sure I'd more questions when I started writing this response mail
 but now I've forgotten those :P

 Thanks,
 Sammy.

 On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
  wrote:

 On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
  Hi list,
 
  I want ot do basic work (add-edit-delete) into asterisk configuration
 files,
  like sip.conf, manager.conf,musiconhold.conf etc.
 
  Please guide me how to configure all these files from from AMI
 connection. I
  am able to login into AMI from Login action but I want to do more
 task in to
  it.
 
  *AMI login:- *
 
  *login.php*
 
  ?php
  $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
  fputs($socket, Action: Login\r\n);
  fputs($socket, UserName: root\r\n);
  fputs($socket, Secret: energy\r\n\r\n);
  ?
  *AMI command:-*
 
  Below commands are for musiconhold.conf. I want to add new MOH
 context into
  it.
  ?php
  include(login.php);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, Filename: musiconhold.conf\r\n);
fputs($socket, Srcfilename: musiconhold.conf\r\n);
fputs($socket, Dstfilename: musiconhold.conf\r\n);
fputs($socket, Action-00: newcat\r\n);
fputs($socket, Cat-00: bhavik\r\n);
fputs($socket, mode: files\r\n);
fputs($socket, directory: /var/lib/asterisk/moh\r\n);
fputs($socket, Reload: yes\r\n);
fputs($socket, ActionID: 9873497149817\r\n);
fputs($socket, Action: Logoff\r\n\r\n);

 You're not really editing. You're writing.

 Note the following:

 * It requires Asterisk to be running, and accessible through the
 manager
  interface.
 * asterisk.conf may be in a path that is not the configuration
  directory. I'm not sure if this special case is handled.
 * #include are basically handled, but mostly for reading. IIRC the
 write
  is back to a single file. No idea about #exec, which will probably
  have odd interactions with UpdateConfig. Configuration templates
  ('[section](template)') are also not handled gracefully.

 
  After doing all no success :((

 This is a report of the the thing that did not happen. Next time you
 ask
 a question

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

Though this might have been resolved/accomplished already but I've couple of
questions for Virendra Bhati.

1- If you are doing this to make new accounts for new users, why couldn't
you use Asterisk realtime(DB) based configurations of
Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
and lots of filing !?

2- Since its a web-based Filing operations and if multiple users are to use
the same page for appending/overwriting their configurations wouldn't it
lead to information being lost when multiple users applying their changes ?
I wonder how do you handle that ?

I'm sure I'd more questions when I started writing this response mail but
now I've forgotten those :P

Thanks,
Sammy.

On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
  Hi list,
 
  I want ot do basic work (add-edit-delete) into asterisk configuration
 files,
  like sip.conf, manager.conf,musiconhold.conf etc.
 
  Please guide me how to configure all these files from from AMI
 connection. I
  am able to login into AMI from Login action but I want to do more task in
 to
  it.
 
  *AMI login:- *
 
  *login.php*
 
  ?php
  $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
  fputs($socket, Action: Login\r\n);
  fputs($socket, UserName: root\r\n);
  fputs($socket, Secret: energy\r\n\r\n);
  ?
  *AMI command:-*
 
  Below commands are for musiconhold.conf. I want to add new MOH context
 into
  it.
  ?php
  include(login.php);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, Filename: musiconhold.conf\r\n);
fputs($socket, Srcfilename: musiconhold.conf\r\n);
fputs($socket, Dstfilename: musiconhold.conf\r\n);
fputs($socket, Action-00: newcat\r\n);
fputs($socket, Cat-00: bhavik\r\n);
fputs($socket, mode: files\r\n);
fputs($socket, directory: /var/lib/asterisk/moh\r\n);
fputs($socket, Reload: yes\r\n);
fputs($socket, ActionID: 9873497149817\r\n);
fputs($socket, Action: Logoff\r\n\r\n);

 You're not really editing. You're writing.

 Note the following:

 * It requires Asterisk to be running, and accessible through the manager
  interface.
 * asterisk.conf may be in a path that is not the configuration
  directory. I'm not sure if this special case is handled.
 * #include are basically handled, but mostly for reading. IIRC the write
  is back to a single file. No idea about #exec, which will probably
  have odd interactions with UpdateConfig. Configuration templates
  ('[section](template)') are also not handled gracefully.

 
  After doing all no success :((

 This is a report of the the thing that did not happen. Next time you ask
 a question, please report what actually does happen (I got the following
 response: ...).

 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

 --
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Re: [asterisk-users] Dialing multiple endpoints and CallerID presentation

[Sam Govind]

Alternative work around to this could be:

1- Make two different dialplan extensions. One to dial DAHDI numbers with
setting for DAHDI and other extension for SIP dialing. Both extensions
setting different CallerID presentation
2- Create a queue with Local extensions as static members (strategy=ringall)

So whenever you want to dial to both B, and C location use the queue
dial-out.

I think it should work.


On Mon, Aug 29, 2011 at 12:15 PM, Olivier oza_4...@yahoo.fr wrote:

 Hi,

 I've got the following use case where I want to simultaneously dial 2
 endpoints that both need different CallerID presentation.
 How can I do that, from the dialplan preferably ?

 For instance, let say phone A needs to both dial B, an internal SIP phone
 and C, a cell phone reachable through a DAHDI span from a an Asterisk system
 where :
 1. users can use 4-digits short numbers to reach other internal phones.
 2. calls going out through the DAHDI span, must have CallerIDs presented
 without any prefix.

 Ideally, CallerID should be presented :
 1- with 4-digits for internal phones
 2- with 10-digits for external phones
 so that both phones can return the call without re-dialing.


 Suggestions ?

 A is 1234 alias DID 051234
 B is 5678
 C is 0123456789
 I was thinking of using something like this:


 Dial(SIP/5678option_to_present_1234_to_calleeDAHDI/g1option_to_present_051234/0123456789)

 What could be option_to_present_1234_to_callee and
 option_to_present_051234

 Regards

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Re: [asterisk-users] Wanted a modified SIP message body

[Sam Govind]

Use the  *SIPAddHeader(Header:Content)* application in dialplan. I don't
think Method specific SIP headers can be done via asterisk.

On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano jaimelozan...@gmail.comwrote:

 Hello everybody,
 I want Asterisk Server to send packets (SIP packets) to some 3Com
 telephones with the text TZ: 7200\n (ie Time Zone = two hours) in the
 message body because 3com PBX sends this variable. I would like to know if I
 it is possible to configure Asterisk to do it, and how.

 have a nice day!

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[asterisk-users] spam blacklist

[Sam]

Just a note, the asterisk mailing list server continually gets 
blacklisted over at 
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering 
mail to spamtraps. Perhaps something needs to be looked into...

Regards,
Sam

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[asterisk-users] Passing parameter from executable program to asterisk dialplan

[Muro, Sam]

I am having a problem understanding the way to retrieve some parameters to
asterisk via AGI or what ever method that fits. I have an executable
program that accept one parameter (CALLERID) and return customer status
from the database server which can be printed in the console.

#./retrive 0117473789
NAME: Franklin John
STATUS: Active

Can someone advice on how i can catch this values from AGI or directly on
dialplan.

Thanks
Sam

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Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

[Muro, Sam]

Kyle Kienapfel wrote:
 On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com
 wrote:
 I am having a problem understanding the way to retrieve some parameters
 to
 asterisk via AGI or what ever method that fits. I have an executable
 program that accept one parameter (CALLERID) and return customer status
 from the database server which can be printed in the console.

 #./retrive 0117473789
 NAME: Franklin John
 STATUS: Active

 Can someone advice on how i can catch this values from AGI or directly
 on
 dialplan.

 Thanks
 Sam

 --

 Hopefully you can modify the executable

 #./retrieve 8675309
 SET VARIABLE name Jenny
 SET VARIABLE status Active

 When running an AGI asterisk expects to have a conversation with the
 application, so when the AGI does a command asterisk reports back with
 whether or not it worked. I know a person can set one variable that
 way, but when I got a need to set two variables I finally broke down
 and read the documentation on AGI's :)

 Start
 Readlines from input until line is blank
 print SET VARIABLE name Jenny
 readline
 print SET VARIABLE status Active
 End

 --

Thanks,
So I you suggesting that the executable to changed to output say
cout SET VARIABLE name Jenny;
and let the AGI retrieve them as per the pseudo you mentioned?

Sam






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Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

[Muro, Sam]

Steve Edwards wrote:
 On Sun, 25 Jul 2010, Muro, Sam wrote:

 I am having a problem understanding the way to retrieve some parameters
 to asterisk via AGI or what ever method that fits. I have an executable
 program that accept one parameter (CALLERID) and return customer status
 from the database server which can be printed in the console.

 #./retrive 0117473789
 NAME: Franklin John
 STATUS: Active

 Can someone advice on how i can catch this values from AGI or directly
 on dialplan.

 AGI is a protocol used to interact with Asterisk. An AGI is a separate
 process created by Asterisk when you execute agi() in the dialplan.

 From best to worst...

 1) You could recode your retrieve application so it uses the AGI protocol.
 Then, you could set channel variables to make these values accessible to
 the rest of your dialplan.

 2) You could cobble up an AGI to execute your retrieve application using a
 pipe (popen() in c), parse the output and set channel variables.

 3) You could cobble up something to execute your retrieve application,
 redirecting the output to a file and then use the FILE function read the
 text file and then parse the output using dialplan functions.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

 --
Thanks Steve
Option one and two looks more ideal. Let stick on option one, the program
is written in C++ (Actually is a corba interface). I have tried looking on
how to write AGI script using C++ in vain. I am used to perl/php for
scripting.. Can you post a snippet of c++ agi script.

Sam

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Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

[Muro, Sam]

Kyle Kienapfel wrote:
 On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam resea...@businesstz.com
 wrote:
 Kyle Kienapfel wrote:
 On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com
 wrote:
 I am having a problem understanding the way to retrieve some
 parameters
 to
 asterisk via AGI or what ever method that fits. I have an executable
 program that accept one parameter (CALLERID) and return customer
 status
 from the database server which can be printed in the console.

 #./retrive 0117473789
 NAME: Franklin John
 STATUS: Active

 Can someone advice on how i can catch this values from AGI or directly
 on
 dialplan.

 Thanks
 Sam

 --

 Hopefully you can modify the executable

 #./retrieve 8675309
 SET VARIABLE name Jenny
 SET VARIABLE status Active

 When running an AGI asterisk expects to have a conversation with the
 application, so when the AGI does a command asterisk reports back with
 whether or not it worked. I know a person can set one variable that
 way, but when I got a need to set two variables I finally broke down
 and read the documentation on AGI's :)

 Start
 Readlines from input until line is blank
 print SET VARIABLE name Jenny
 readline
 print SET VARIABLE status Active
 End

 --

 Thanks,
 So I you suggesting that the executable to changed to output say
 cout SET VARIABLE name Jenny;
 and let the AGI retrieve them as per the pseudo you mentioned?

 Sam


 Does doing that output a newline at the end of the line?

 If it doesn't you might want something more like (i am just guessing
 syntax here btw)
 cout SET VARIABLE name Jenny  ENDL;
 or
 cout SET VARIABLE name Jenny\n;

 --
You are right. Both of them are correct syntax
I will give it a try and revert

Sam

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Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

[Muro, Sam]

Steve Edwards wrote:
 On Sun, 25 Jul 2010, Muro, Sam wrote:

 I am having a problem understanding the way to retrieve some
 parameters to asterisk via AGI or what ever method that fits. I have
 an executable program that accept one parameter (CALLERID) and return
 customer status from the database server which can be printed in the
 console.

 #./retrive 0117473789
 NAME: Franklin John
 STATUS: Active

 Can someone advice on how i can catch this values from AGI or directly
 on dialplan.

 Steve Edwards wrote:

 AGI is a protocol used to interact with Asterisk. An AGI is a separate
 process created by Asterisk when you execute agi() in the dialplan.

 From best to worst...

 1) You could recode your retrieve application so it uses the AGI
 protocol. Then, you could set channel variables to make these values
 accessible to the rest of your dialplan.

 2) You could cobble up an AGI to execute your retrieve application
 using a pipe (popen() in c), parse the output and set channel
 variables.

 3) You could cobble up something to execute your retrieve application,
 redirecting the output to a file and then use the FILE function read
 the text file and then parse the output using dialplan functions.

 On Sun, 25 Jul 2010, Muro, Sam wrote:

 Option one and two looks more ideal. Let stick on option one, the
 program is written in C++ (Actually is a corba interface). I have tried
 looking on how to write AGI script using C++ in vain. I am used to
 perl/php for scripting.. Can you post a snippet of c++ agi script.

 I'm a c weenie myself. I coded my own library way too long ago and
 remember the scars :)

 If you google for asterisk agi c++ library you'll find links to cagi,
 quivr, and probably a couple more. I don't know if any are c++ specific.

 The basic outline is:

   call a function to read the AGI environment from stdin. (Mine is
   named agi_read_environment())

   get your customer ID number either from the command line or from a
   channel variable. (argv[] or agi_get_variable(CUSTOMER-ID,
   customer_id.)

   retrieve your values from your database.

   set your channel variables. (agi_set_variable(CUSTOMER-NAME,
   mysql_row[CUSTOMER_NAME]))

 --
Thanks Steve

Let me check them out and give them a try. I really appreciate your input

Sam

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[asterisk-users] Corba interface

[Muro, Sam]

Hi there
Has anyone know how to configure asterisk to be able to query Corba
interface directly from the dialplan

Sam

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[asterisk-users] Corba interface

[Muro, Sam]

Hi there

Does anyone know how to configure asterisk to be able to query Corba
interface directly from the dialplan

Sam



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Re: [asterisk-users] Using asterisk as avaya definity recordingserver

[Muro, Sam]

Moises Silva wrote:
 On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi
 pr...@practis.com.brwrote:

 Hi, it's not that simple.
 It requires deep modification on asterisk and dahdi sources to work the
 way
 you want.


 Why? I must confess I still don't quite understand what he wants, from
 what
 I've read the legacy pbx will place a secondary call via ISDN ( did he
 mean
 PRI? ) therefore Asterisk will just Record(), what is it that is not so
 simple about that?

Hi Moses
Task: Recording phone calls

Here is the scenario;
- A legacy system is connected back to back to asterisk pbx with PRI
connection and asterisk is connected to the telco via PRI
Users(Analog/Digital) Legacy
(PRI)-Asterisk---(PRI)---Telco
- Telco to users (vise versa) need to be recorded on asterisk - Easily Done
- Internal calls (extension to extension) on legacy need to be recording
(currently is done via Nice) on asterisk - This's the problem

Sam


 --
 Moises Silva
 Senior Software Engineer
 Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
 9T3
 Canada
 t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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Re: [asterisk-users] Using asterisk as avaya definity recording server

[Muro, Sam]

Hi there

Looks like someone hasnt done this!! I have been thinking and find out
that Monitor/Spy and the likes wont help me as the call need to be bridged
with the asterisk core or via channel drivers.

My final shot now is on Record() function. Since the legacy system will
forward the call to the monitoring interfaces when bridged within itself,
it the interface in on Asterisk, then we can capture the pattern and use

exten =
#CALLER_NUMBER#CALLED_NUMBER,1,Record(/var/spool/asterisk/monitor/avaya-${EXTEN:1:4}-${EXTEN:4:4}:wav)

This assume that Len(CALLER_NUMBER) = 4

Anyone with alternative solution?

Muro, Sam wrote:
 Oh.. I didnt know that.

 Thanks
 Sam
 Muro, Sam escribió:
 What do you mean chief? What am looking at is ability for asterisk to
 receive a call and recording until it tier down without bridging it to
 the
 physical device

 Sam

 Would you like the advice in all caps?


 He means that you put the subject in all caps. He normally gets upset
 with everyone that does this on the subject or in the body. I've
 corrected the caps in the subject to avoid further upsetting.

 Cheers,




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Re: [asterisk-users] asterisk-users Digest, Vol 68, Issue 33

[Muro, Sam]

What do you mean chief? What am looking at is ability for asterisk to
receive a call and recording until it tier down without bridging it to the
physical device

Sam
 Would you like the advice in all caps?

 On 03/15/2010 01:20 AM, RESEARCH wrote:

 Hi there

 I remember to ask this question in the past but now I have thought of
 something little bit difference. While I understand that asterisk
 dialplan
 accept the call to be answered[ Answer() ] in the dialplan, I wanna know
 if
 this is possible;
 i. A call on legacy PBX, extension to extension is made.
 ii. On call bridging, the legacy PBX initiate a third bridging to the
 recording system via an ISDN interface.
 iii. Conversation on Legacy continue but asterisk record this call until
 hangup is issued

 Please advice if this is possible.

 Sam




 --
 Alex Balashov - Principal
 Evariste Systems LLC

 Tel: +1 678-954-0670
 Direct : +1 678-954-0671
 Web: http://www.evaristesys.com/


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Re: [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER

[Muro, Sam]

What do you mean chief? What am looking at is ability for asterisk to
receive a call and recording until it tier down without bridging it to the
physical device

Sam
 Would you like the advice in all caps?

 On 03/15/2010 01:20 AM, RESEARCH wrote:

 Hi there

 I remember to ask this question in the past but now I have thought of
something little bit difference. While I understand that asterisk
dialplan
 accept the call to be answered[ Answer() ] in the dialplan, I wanna
know if
 this is possible;
 i. A call on legacy PBX, extension to extension is made.
 ii. On call bridging, the legacy PBX initiate a third bridging to the
recording system via an ISDN interface.
 iii. Conversation on Legacy continue but asterisk record this call
until hangup is issued

 Please advice if this is possible.

 Sam




 --
 Alex Balashov - Principal
 Evariste Systems LLC

 Tel: +1 678-954-0670
 Direct : +1 678-954-0671
 Web: http://www.evaristesys.com/




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Re: [asterisk-users] Using asterisk as avaya definity recording server

[Muro, Sam]

Oh.. I didnt know that.

Thanks
Sam
 Muro, Sam escribió:
 What do you mean chief? What am looking at is ability for asterisk to
 receive a call and recording until it tier down without bridging it to the
 physical device

 Sam

 Would you like the advice in all caps?


 He means that you put the subject in all caps. He normally gets upset
 with everyone that does this on the subject or in the body. I've
 corrected the caps in the subject to avoid further upsetting.

 Cheers,


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Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

[Muro, Sam]

Hi Steve

 Even though you shouldn't have to, have your rebooted?  200 days of
 uptime and this just started?
It seems this problem is common as i have three boxes of the same capacity
with exactly the same problem. So reboot should only solve the problem for
a while


 Have you recently updated the box?

No.

 ksoftirqd seems to have issues in some kernels.  That is where I would
 start after restarting Asterisk and or the server.


Allow me to look at it and revert

 http://tinyurl.com/ygd2eha

 Thanks,
 Steve T





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Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

[Muro, Sam]

 Hi Team

 Can someone advice me on how i can lower the load average on my asterisk
 server?

 dahdi-linux-2.1.0.4
 dahdi-tools-2.1.0.2
 libpri-1.4.10.1
 asterisk-1.4.25.1

 2 X TE412P Digium cards on ISDN PRI

 Im using the system as an IVR without any transcoding or bridging

 **
 top - 10:27:57 up 199 days,  5:18,  2 users,  load average: 67.75,
 62.55,
 55.75
 Tasks: 149 total,   1 running, 148 sleeping,   0 stopped,   0 zombie
 Cpu0
 : 10.3%us, 32.0%sy,  0.0%ni, 57.3%id,  0.0%wa,  0.0%hi,  0.3%si,  0.0%st
 Cpu1  : 10.6%us, 34.6%sy,  0.0%ni, 54.8%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
 Cpu2  : 13.3%us, 36.5%sy,  0.0%ni, 49.8%id,  0.0%wa,  0.0%hi,  0.3%si,
 0.0%st
 Cpu3  :  8.6%us, 39.5%sy,  0.0%ni, 51.8%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
 Cpu4  :  7.3%us, 38.0%sy,  0.0%ni, 54.7%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
 Cpu5  : 17.9%us, 37.5%sy,  0.0%ni, 44.5%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
 Cpu6  : 13.3%us, 37.2%sy,  0.0%ni, 49.5%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
 Cpu7  : 12.7%us, 37.3%sy,  0.0%ni, 50.0%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st

 System is fairly loaded, but there's still plenty of idle CPU cycles. If
 we were in a storm of CPU-intensive processes, we would have expected
 many more running processes. Right now we have none (the single
 process is 'top' itself).

 Mem:   3961100k total,  3837920k used,   123180k free,   108944k buffers
 Swap:   779144k total,   56k used,   779088k free,  3602540k cached

   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND 683
 root  15   0 97968  36m 5616 S 307.7  0.9  41457:34 asterisk
 17176 root  15   0  2196 1052  800 R  0.7  0.0   0:00.32 top
 1 root  15   0  2064  592  512 S  0.0  0.0   0:13.96 init
 2 root  RT  -5 000 S  0.0  0.0   5:27.80 migration/0
 3

 Processes seem to be sorted by size. You should have pressed 'p' to go
 back to sorting by CPU. Now we don't even see the worst offenders.

Tried option 'p' but doesnt seems to exist. Centos 5.3 kernel 2.6.18-128


 root  34  19 000 S  0.0  0.0   0:00.11 ksoftirqd/0 4
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/0 5
 root  RT  -5 000 S  0.0  0.0   1:07.67 migration/1 6
 root  34  19 000 S  0.0  0.0   0:00.09 ksoftirqd/1 7
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/1 8
 root  RT  -5 000 S  0.0  0.0   1:16.92 migration/2 9
 root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/2
10 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/2
 11
 root  RT  -5 000 S  0.0  0.0   1:34.54 migration/3 12
 root  34  19 000 S  0.0  0.0   0:00.15 ksoftirqd/3 13
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/3 14
 root  RT  -5 000 S  0.0  0.0   0:54.66 migration/4 15
 root  34  19 000 S  0.0  0.0   0:00.01 ksoftirqd/4 16
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/4 17
 root  RT  -5 000 S  0.0  0.0   1:39.64 migration/5 18
 root  39  19 000 S  0.0  0.0   0:00.21 ksoftirqd/5 19
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/5 20
 root  RT  -5 000 S  0.0  0.0   1:06.27 migration/6 21
 root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/6 22
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/6 23
 root  RT  -5 000 S  0.0  0.0   1:23.24 migration/7 24
 root  34  19 000 S  0.0  0.0   0:00.17 ksoftirqd/7 25
 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/7 26
 root  10  -5 000 S  0.0  0.0   0:25.70 events/0 27 root
  10  -5 000 S  0.0  0.0   0:37.83 events/1 28 root
 10  -5 000 S  0.0  0.0   0:15.67 events/2 29 root  10
 -5 000 S  0.0  0.0   0:40.36 events/3 30 root  10  -5
   000 S  0.0  0.0   0:16.45 events/4

 Those are all kernel threads rather than real processes.

 So I suspect one of two things:

 1. You're right after such a storm. The load average will decreases
 sharply.
What do you mean Trafrir

Its obvious that the effect increases with increase number of active
channels. e.g. @channels=90, load average = 4 but @channels =235, load
average= 60+

 2. There are many processes hung in state 'D' (uninterruptable system
 call). If a process is hung in such a system call for long, it normally
 means a problem. E.g. disk-access issues which causes all processes
 trying to acess a certain file to hang.

I presume this should happen if there is irq sharing between disks and
cards which isnt my case.

 --


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Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

[Muro, Sam]

 Hi Team

 Can someone advice me on how i can lower the load average on my
asterisk server?

 dahdi-linux-2.1.0.4
 dahdi-tools-2.1.0.2
 libpri-1.4.10.1
 asterisk-1.4.25.1

 2 X TE412P Digium cards on ISDN PRI

 Im using the system as an IVR without any transcoding or bridging

 **
 top - 10:27:57 up 199 days,  5:18,  2 users,  load average: 67.75, 62.55,
 55.75


 Hi Sam!

Hello Steve!

 Are there any side-effects from the high load average?  The system
doesn't seem to be CPU or disk bound from the look of the CPU stats. 
System %age is
 high by way - software echo cancellaton?, and Asterisk is using a lot of
cpu
 which isn't suprising.

Yes. Audio quality issues. I have enabled the hardware echo cancellation
and configured
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes


 I'm guessing you are running 8 spans and 200+ calls into your IVR?


You are correct. 8 span which process up to 240 calls at pick time

 If the system is actually performing fine then I'd just say that there
is something about the Asterisk threads that makes them look runnable
and that
 accounts for the high load average.  Is the IVR an agi or fastagi or
what? -

I have the agi scripts not as ivr but to help populate the required
information into mysql db. Probably here is where the problem lies i have
to connect and disconnect to mysql each time a call is made or a specific
menu is selected

Here is the script
*
#!/usr/bin/perl -w
use strict;
use DBI();
use Scalar::Util qw/weaken/;

my $cdr_log_file = /var/log/asterisk/ivr_log;
my $mysql_host = cdr01;
my $mysql_db = ivrcdrdb;
my $mysql_table = tbl_ivrcdr_details;
my $mysql_user = ivruser;
my $mysql_pwd = a09876a;


my $sth;

my $data0= $ARGV[0];
my $data1= $ARGV[1];
my $data2= $ARGV[2];
my $data3= $ARGV[3];
my $data4= $ARGV[4];
my $data5= $ARGV[5];
my $data6= $ARGV[6];
my $data7= $ARGV[7];


# Connect to database
# print Connecting to database...\n\n;
my $dbh =
DBI-connect(DBI:mysql:database=$mysql_db;host=$mysql_host,$mysql_user,$mysql_pwd,{'RaiseError'
= 1});

my $insert_str = insert into $mysql_table (calldate, language, src,
duration, accountcode, uniqueid, currentmenu, nextmenu) values
(\$data0\, \$data1\, \$data2\, \$data3\,  \$data4\, \$data5\,
\$data6\, \$data7\);\n;
   $sth = $dbh-prepare($insert_str);
   $sth-execute();

# print \n\nOK.\n;

$sth-finish();
$dbh-disconnect();


# Trying to resolve memory leak should it happen
delete($ARGV[0]);
delete($ARGV[1]);
delete($ARGV[2]);
delete($ARGV[3]);
delete($ARGV[4]);
delete($ARGV[5]);
delete($ARGV[6]);
delete($ARGV[7]);


exit;
*

 the code path may have a spinlock logic to it that means that many
threads
 are runnable but when scheduled just go back to sleep.  That would
account for high load average with lots of spare CPU.  If that's what is
happening then I wouldn't worry much more about it.

 Regards,
 Steve

Regards
Sam



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[asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

[Muro, Sam]

Hi Team

Can someone advice me on how i can lower the load average on my asterisk
server?

dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1

2 X TE412P Digium cards on ISDN PRI

Im using the system as an IVR without any transcoding or bridging

**
top - 10:27:57 up 199 days,  5:18,  2 users,  load average: 67.75, 62.55,
55.75
Tasks: 149 total,   1 running, 148 sleeping,   0 stopped,   0 zombie Cpu0 
: 10.3%us, 32.0%sy,  0.0%ni, 57.3%id,  0.0%wa,  0.0%hi,  0.3%si,  0.0%st
Cpu1  : 10.6%us, 34.6%sy,  0.0%ni, 54.8%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Cpu2  : 13.3%us, 36.5%sy,  0.0%ni, 49.8%id,  0.0%wa,  0.0%hi,  0.3%si, 
0.0%st
Cpu3  :  8.6%us, 39.5%sy,  0.0%ni, 51.8%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Cpu4  :  7.3%us, 38.0%sy,  0.0%ni, 54.7%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Cpu5  : 17.9%us, 37.5%sy,  0.0%ni, 44.5%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Cpu6  : 13.3%us, 37.2%sy,  0.0%ni, 49.5%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Cpu7  : 12.7%us, 37.3%sy,  0.0%ni, 50.0%id,  0.0%wa,  0.0%hi,  0.0%si, 
0.0%st
Mem:   3961100k total,  3837920k used,   123180k free,   108944k buffers
Swap:   779144k total,   56k used,   779088k free,  3602540k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND 683
root  15   0 97968  36m 5616 S 307.7  0.9  41457:34 asterisk
17176 root  15   0  2196 1052  800 R  0.7  0.0   0:00.32 top
1 root  15   0  2064  592  512 S  0.0  0.0   0:13.96 init
2 root  RT  -5 000 S  0.0  0.0   5:27.80 migration/0 3
root  34  19 000 S  0.0  0.0   0:00.11 ksoftirqd/0 4
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/0 5
root  RT  -5 000 S  0.0  0.0   1:07.67 migration/1 6
root  34  19 000 S  0.0  0.0   0:00.09 ksoftirqd/1 7
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/1 8
root  RT  -5 000 S  0.0  0.0   1:16.92 migration/2 9
root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/2
   10 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/2 11
root  RT  -5 000 S  0.0  0.0   1:34.54 migration/3 12
root  34  19 000 S  0.0  0.0   0:00.15 ksoftirqd/3 13
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/3 14
root  RT  -5 000 S  0.0  0.0   0:54.66 migration/4 15
root  34  19 000 S  0.0  0.0   0:00.01 ksoftirqd/4 16
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/4 17
root  RT  -5 000 S  0.0  0.0   1:39.64 migration/5 18
root  39  19 000 S  0.0  0.0   0:00.21 ksoftirqd/5 19
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/5 20
root  RT  -5 000 S  0.0  0.0   1:06.27 migration/6 21
root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/6 22
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/6 23
root  RT  -5 000 S  0.0  0.0   1:23.24 migration/7 24
root  34  19 000 S  0.0  0.0   0:00.17 ksoftirqd/7 25
root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/7 26
root  10  -5 000 S  0.0  0.0   0:25.70 events/0 27 root
 10  -5 000 S  0.0  0.0   0:37.83 events/1 28 root 
10  -5 000 S  0.0  0.0   0:15.67 events/2 29 root  10 
-5 000 S  0.0  0.0   0:40.36 events/3 30 root  10  -5  
  000 S  0.0  0.0   0:16.45 events/4
*

Thanks
Sam



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