[asterisk-users] Asterisk, Sent accountcode between 2 asterisk

[Olivier CALVANO]

Hi

I have two Asterisk Server:

The first server "A", all phone are connected
The Second server "B" only route call to a lot of SIP supplier

the server A sent:

; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten => _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten => _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten => _X.,3,Hangup


anyone know if it's possible to add the CDR Accountcode to this process
for get it on the second server "B" ?

i want the same accountcode on the 2 servers

thanks
Olivier

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[asterisk-users] Help Asterisk / API / Perl

[Olivier CALVANO]

Hi

i want use the API on my asterisk 1.6, but i have a small problems :

In extension, i start it :
exten => _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems

but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)

but that's don't work:

use Asterisk::AGI;
use lib "/var/lib/asterisk/agi-bin";
$AGI = new Asterisk::AGI;
$typ = $AGI->get_variable('agi_type');

$typ don't have SIP or IAX, same test without succes:
$typ = $AGI->get_variable('type');

anyone know this problems ?

thanks
Olivier

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Re: [asterisk-users] TLS/SRTP calls go to circuit busy.

[Mitch Johnson]

> Once again, thanks for your reply.  I had done some research already but 
> forget to include it in my previous email.  I did find a bug that is 
> remarkably similar to the issues that I'm having.  The bug number is 18674.

Thanks,

Mitch Johnson

> Message: 8
> Date: Fri, 04 Mar 2011 00:34:45 -0600
> From: Terry Wilson 
> Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Message-ID: <4d708805.3060...@digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> On 03/03/2011 02:22 PM, Mitch Johnson wrote:
>> Thanks so much for pointing this out.  I was curious why the commands in the 
>> documentation differed to the commands I was using.
>> 
>> That problem is fixed, but now I have a new issue.  I can call with no 
>> issues, however, as soon as I answer one of the calls I see the error: 
>> ast_srtp_unprotect:  SRTP unprotect: authentication failure.  Below is a 
>> snippet of the debug as the call is answered.
> The best thing to do at this point would be to file a bug report with 
> the info at which point it will eventually probably be assigned to me 
> (unless some awesome person comes up with a fix first!) to look at. If I 
> have a bit of free time, I'll try to take a peek at it. If you can post 
> the sip debug output of the entire offer/answer exchange to the bug 
> report, it will help greatly.
> 
> Terry
> 


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Re: [asterisk-users] can anyone tell me how to set asterisk to record all phonecall

[Jeremy Kister]

On 3/4/2011 9:49 PM, John Wu wrote:

I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.


this is one way it can be done

make sure you have 'lame' installed.

- in your extensions.conf:

[global]
VSA=/var/spool/asterisk

[outbound-or-wherever-you-dial]
exten => _XXX,1,Macro(Snoop,${EXTEN})
exten => _XXX,n,Dial(SIP/${EXTEN},${TIMEOUT})
exten => _XXX,n,StopMixMonitor
; above in case you're in some loop & Dial fails,
; e.g., swift+monitor crash asterisk


[macro-Snoop]
; ${ARG1} channel
exten => s,1,GotoIf($["${SNOOPING}" = "1"]?snooping)
exten => s,n,Set(SNOOPING=1)
exten => s,n,Set(=${STRFTIME(${EPOCH},,%Y)})
exten => s,n,Set(MM=${STRFTIME(${EPOCH},,%m)})
exten => s,n,Set(DD=${STRFTIME(${EPOCH},,%d)})
exten => s,n,Set(HMS=${STRFTIME(${EPOCH},,%H%M%S)})
exten => s,n,Set(FILENAME=${HMS}-${CALLERID(num)}-${ARG1}-${UNIQUEID})
exten => s,n,Set(MIXMON_ARGS=mkdir -p ${VSA}/monitor/${}/${MM}/${DD} 
&& nice -n 19 /usr/local/bin/lame --silent --resample 11.025 -b 16 -t -m 
m ${VSA}/monitor/${FILENAME}.wav 
${VSA}/monitor/${}/${MM}/${DD}/${FILENAME}.mp3 && rm -f 
${VSA}/monitor/${FILENAME}.wav)

exten => s,n,MixMonitor(${FILENAME}.wav,,${MIXMON_ARGS})
exten => s,n(snooping),NoOp(snooping on ${CHANNEL})



that'll end up putting a mp3 of the call in 
/var/spool/asterisk/monitor//MM/DD/HHMMSS-CALLERID.mp3


don't forget any legal issues you might have to work around, recording 
the fact that you declared the message is being recorded.



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http://jeremy.kister.net./

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[asterisk-users] [announce] jkSMS

[Jeremy Kister]

For those interested, I have released a first version of jkSMS, which is 
a simple package that lets cell phones text messages to "asterisk".


Note it's not real SMS, it makes heavy use of email-to-sms gateways, but 
it seems to work well.  I have had the code running > 12 hours, but 
haven't found any issues.


it's not for the faint-of-heart and might require a bit of hacking 
(really minimal though) if you're not running the same tools that i'm 
running (like editing the code's DSN if you dont have sqlite installed)


http://jeremy.kister.net/code/asterisk/jkSMS/

enjoy,

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http://jeremy.kister.net./

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[asterisk-users] can anyone tell me how to set asterisk to record all phonecall

[John Wu]

Hi all,
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
Now I need to set the configure file to let asterisk auto record all
calls. I have searched many
document but still can not succeed. My version is 1.8beta and I prefer
using mixmonitor.

Regards!

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Re: [asterisk-users] How is Libpri developped ?

[Moises Silva]

On Fri, Mar 4, 2011 at 12:50 AM, Olivier  wrote:

> Hi,
>
> Can you explain the main differences between Libpri 1.4.11 and 1.4.12 as
> both seem to receive additions and patches ?

Do they target different asterisk versions ?
> Can they both be considered as production-ready ?
>
>
1.4.12 is just a newer version than 1.4.11 and any released version is as
production-ready as can be reasonably be expected AFAIK.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com
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Re: [asterisk-users] server performance....

[Sevana Oy]

Hi,

We have worked out another approach for load testing:

- generate using sipp certain number of test calls and that go to PBX echo 
server playing and receiving back pre-defined audio

- generate +1 test call, which also plays and receives back an audio file

Then we test the audio we received from the +1 test call using AQuA (Audio 
Quality Analyzer) and obtain a MOS score (AQuA is doing perceptual audio 
quality assessment, it's not calculating MOS as in G.107, but more likely in 
P.862, although the algorithms are absolutely different).


In this way we can always know how many calls can the PBX under test handle 
before actual call quality goes down. The whole test suit is put together 
with other testing (loop back call testing, conference bridge testing) 
capabilities into what we call Asterisk VQM. If my previous message goes 
through moderation you will be able to see screenshots as well :)


Best Regards,
Sevana Oy
http://www.sevana.fi


- Original Message - 
From: "Andrew Latham" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Friday, March 04, 2011 8:20 PM
Subject: Re: [asterisk-users] server performance



On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
 wrote:

Hi every one,
I am doing some experiments on asterisk server
performance.. How can we know server performance? can any one explain 
me

plz
I have 2 doubts regarding the asterisk server performance...

1. When can we know asterisk server performance?
1. when server is in idle state ?
2. when the server is in busy state?

can any one please tell me when can the server performance is known i 
mean

when server is busy or in idle state?

Best Regards,
viswavardhanreddy



Many people test their servers with call-setups and call tear-downs.
Using another tool like sipp you can send 100-1000s of call-setups and
then do call tear-downs.  You can also use transcoding loops to test
the load.  If you have 1 call that is sent to a context where it dials
exten+1 and continues the loop until a target number, you can then set
the codec for each dialed number.  I know that there are many methods
of testing and this is just a common one.

~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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[asterisk-users] 2 ip phones and 1 normal, can't neither send nor receive calls at all...

[Francisco Javier Cintrón Olguín]

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco
spa8800, all them are internal lines.

1.- spa921, 401 ext
2.- spa921, 402 ext
3.- normal phone connected to spa8800 404 ext.

It had a very strange behavior when I was configuring call transfer and call
pickup.

These are steps to repeat it:


1.- from 401 call to 404
2.- from 404 don't answer it.
3.- from 402 press *8 and wait 10 seconds
4.- 402 says that it is connected.
5.- 404 stops to sound.
6.- 401 keeps ringing
7.- Hang up 402
8.- Hang up 401


After these steps I can not neither send nor receive calls from anyone of
401, 402 or 404 until I restart asterisk.

/var/log/asterisk/messages, doesn´t show anything strange.

¿what's happening with my phones?

Thank you for your kind help.
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Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?

[Adrian Serafini]

Hi,

We use Opensips and like the results.  The forks are similar, docs from 
one can help in the other.  The opensips mailing list is monitored by 
one of the main developers.  He is even in the IRC chat in the mornings.


The docs are kept current on the opensips webpage.  They like to change 
modules a bit, so really watch your versions.  The commercial PDF 
"Building Telephony Systems with OpenSIPS 1.6" is excellent.(duck)


Yum is nice for the dependencies, but I would use a compile for 
Opensips.  Most of the docs are Debian specific.  I love Debian, but our 
clients love Centos.  I have some Centos Opensips compile docs if needed.


There are a few GUI's, but I prefer Opensips-cp.  To put opensips-cp on 
a remote server, you need the xmlrpc module loaded on opensips.  This 
works in Debian but fails on Centos (64 bit ONLY).


Good luck,

Adrian



On 03/04/2011 01:49 PM, Steve Edwards wrote:

I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.

Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates.

I'm leaning towards OpenSIPS because it's in EPEL so I can install it
with yum. Also, because I think the name sounds more 'professional' when
discussing architecture with clients :)

Which do you use and why?



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Re: [asterisk-users] chan_skinny and Cisco 793X (7936) support in1.8

[Danny Nicholas]

Skinny?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alfred
Monticello
Sent: Friday, March 04, 2011 2:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_skinny and Cisco 793X (7936) support
in1.8



Does anybody have an answer to this?



- Original Message 
From: Alfred Monticello 
To: asterisk-users@lists.digium.com
Sent: Wed, March 2, 2011 9:59:20 PM
Subject: chan_skinny and Cisco 793X (7936) support in 1.8


Is there any way to make a Cisco 7936 conference phone work in version 1.8?


  

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Re: [asterisk-users] chan_skinny and Cisco 793X (7936) support in 1.8

[Alfred Monticello]

Does anybody have an answer to this?



- Original Message 
From: Alfred Monticello 
To: asterisk-users@lists.digium.com
Sent: Wed, March 2, 2011 9:59:20 PM
Subject: chan_skinny and Cisco 793X (7936) support in 1.8


Is there any way to make a Cisco 7936 conference phone work in version 1.8?


  

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Re: [asterisk-users] server performance....

[Warren Selby]

On Fri, Mar 4, 2011 at 11:28 AM, viswavardhanreddy karna <
viswavardhanre...@gmail.com> wrote:

> Hi,
>I mean when the cpu history is in idel and in busy state...
>
> i have one more doubt that we are doing experiments on server
> performance(only on software) it does not depends on hardware or even on
> systemm/...
>
> knowing the server performance only the software  side includes any cpu
> history like when the server is busy or idle
>
>
Have a look at munin, or maybe cacti or even mrtg.

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?

[Amit Nepal]

Hi,
  I have been working on a project with asterisk and kamailio. I would 
prefer using kamailio because i have personally met with the developers 
and it has more active users and rapid developments. The developers are 
also very friendly and helpful. And well open ser is not gone, the name 
is changed to kamailio  I guess. It had a fork, but now they have merged 
together.


Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
   602-234-0917#112
http://www.phoenixinternet.net


On 3/4/2011 11:49 AM, Steve Edwards wrote:
I'm starting a new project similar to a previous project where I used 
OpenSER to front a bunch of Asterisk servers.


Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely 
candidates.


I'm leaning towards OpenSIPS because it's in EPEL so I can install it 
with yum. Also, because I think the name sounds more 'professional' 
when discussing architecture with clients :)


Which do you use and why?



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[asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?

[Steve Edwards]

I'm starting a new project similar to a previous project where I used 
OpenSER to front a bunch of Asterisk servers.


Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely 
candidates.


I'm leaning towards OpenSIPS because it's in EPEL so I can install it with 
yum. Also, because I think the name sounds more 'professional' when 
discussing architecture with clients :)


Which do you use and why?

--
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] Asterisk <-> Lync / Call Center Transfer / Refer

[Louis Carreiro]

Ha! Thanks Vip!

Sorry about not including my version numbers too. On my production box I'm 
using 1.8.3 (that's the debug from the original email). On my demo box I just 
build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm 
not sure if this is a chan_sip.c problem or if this is a dial plan problem.

So digging in a bit deeper, Asterisk is receving the real REFER message. The 
"REFER-TO: 
"
 is accurate and in chan_sip.c it knows how to manipulate it. It does grab the 
"from-tag" and "to-tag" and parses the data.  On one of the lines below you can 
see it says "Looking for  Call ID: 
655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 
15826bef52 --To-tag as41bacc0b". Then it moves on to bridging the 
peers/channels together. It's not until later that I get the final " SIP/2.0 
481 Call leg/transaction does not exist" which doesn't make sense to me. Also, 
the Lync client says "Call was not transferred because [Original Extension] 
cannot be reached and may be offline."

<->
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  0 [ 53]: REFER 
sip:1820@10.10.10.10:5060;transport=TCP SIP/2.0
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  1 [ 78]: FROM: 
;epid=E5790B0758;tag=15826bef52
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  2 [ 41]: TO: 
;tag=as41bacc0b
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  3 [ 13]: CSEQ: 2 REFER
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  4 [ 58]: CALL-ID: 
655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  5 [ 16]: MAX-FORWARDS: 70
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  6 [ 59]: VIA: SIP/2.0/TCP 
20.20.20.20:5068;branch=z9hG4bK70e8a145
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  7 [107]: CONTACT: 

[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  8 [ 17]: CONTENT-LENGTH: 0
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header  9 [200]: REFER-TO: 

[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Header 10 [ 40]: USER-AGENT: 
RTCC/4.0.0.0 MediationServer
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: --- (11 headers 0 lines) ---
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: = Looking for  Call ID: 
655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 
15826bef52 --To-tag as41bacc0b
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c:  Received REFER (9) - Command in 
SIP REFER
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: Call 
655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 got a SIP call transfer from 
caller: (REFER)!
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: Attended transfer: Will use 
Replace-Call-ID : a9b5f241-5e9d-4439-b347-2cac9384a627 F-tag: aa19f11d4f T-tag: 
7a9abe27a5
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: SIP transfer to extension 
lyncserver.internal.name:5068@from-internal-xfer by (null)
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: SIP attended transfer: Transferer 
channel SIP/Lync-0003, transferee channel SIP/1820-0002
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: Got SIP transfer, applying to 
bridged peer 'SIP/1820-0002'
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c:
<--- Transmitting (no NAT) to 20.20.20.20:5068 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/TCP 20.20.20.20:5068;branch=z9hG4bK70e8a145;received=20.20.20.20
From: ;epid=E5790B0758;tag=15826bef52
To: ;tag=as41bacc0b
Call-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060
CSeq: 2 REFER
Server: FPBX-2.8.1(1.8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Contact: 
Content-Length: 0


<>
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: Trying to put 'SIP/2.0 202' onto TCP 
socket destined for 20.20.20.20:5068
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: Looking for callid 
a9b5f241-5e9d-4439-b347-2cac9384a627 (fromtag aa19f11d4f totag 7a9abe27a5)
[Mar  4 12:54:53] DEBUG[11296] chan_sip.c: Strict routing enforced for session 
655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: set_destination: Parsing 
 for 
address/port to send to
[Mar  4 12:54:53] DEBUG[11296] netsock2.c: Splitting 
'lyncserver.internal.name:5068' gives...
[Mar  4 12:54:53] DEBUG[11296] netsock2.c: ...host 'lyncserver.internal.name' 
and port '5068'.
[Mar  4 12:54:53] DEBUG[11293] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1820-0002
Variable: SIPREFERRINGCONTEXT
Value: from-internal
Uniqueid: 1299261284.2


[Mar  4 12:54:53] DEBUG[11293] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1820-0002
Variable: SIPREFERREDBYHDR
Value:
Uniqueid: 1299261284.2


[Mar  4 12:54:53] DEBUG[11296] netsock2.c: Splitting '20.20.20.20' gives...
[Mar  4 12:54:53] DEBUG[11296] netsock2.c: ...host '20.20.20.20' and port 
'(null)'.
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: set_destination: set destination 
to 20.20.20.20:5068
[Mar  4 12:54:53] VERBOSE[11296] chan_sip.c: Reliably Transmittin

Re: [asterisk-users] server performance....

[viswavardhanreddy karna]

Hi,
   I mean when the cpu history is in idel and in busy state...

i have one more doubt that we are doing experiments on server
performance(only on software) it does not depends on hardware or even on
systemm/...

knowing the server performance only the software  side includes any cpu
history like when the server is busy or idle





BR,
viswavardhan

On Fri, Mar 4, 2011 at 6:25 PM, viswavardhanreddy karna <
viswavardhanre...@gmail.com> wrote:

> HI,
>  The way you said is correct, we are using SIPp to generate as many
> calls as it can send and and the server is able is to take simultaneously of
> 560 - 570 calls
>
> 1. when we kept server for some time as idle it took 575 calls
> 2. when we kept again server as busy by continous calls back to back it is
> taking 560-570 between i am not knowing which boundary should i take in
> this
>
> should i take the boundary of max successfull calls  when server is in busy
> state or when server is in idle state?
>
>
>
>
> Best Regards,
> viswavardhan
>
>
> On Fri, Mar 4, 2011 at 6:20 PM, Andrew Latham  wrote:
>
>> On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
>>  wrote:
>> > Hi every one,
>> >  I am doing some experiments on asterisk server
>> > performance.. How can we know server performance? can any one
>> explain me
>> > plz
>> >  I have 2 doubts regarding the asterisk server performance...
>> >
>> > 1. When can we know asterisk server performance?
>> > 1. when server is in idle state ?
>> > 2. when the server is in busy state?
>> >
>> > can any one please tell me when can the server performance is known i
>> mean
>> > when server is busy or in idle state?
>> >
>> > Best Regards,
>> > viswavardhanreddy
>>
>>
>> Many people test their servers with call-setups and call tear-downs.
>> Using another tool like sipp you can send 100-1000s of call-setups and
>> then do call tear-downs.  You can also use transcoding loops to test
>> the load.  If you have 1 call that is sent to a context where it dials
>> exten+1 and continues the loop until a target number, you can then set
>> the codec for each dialed number.  I know that there are many methods
>> of testing and this is just a common one.
>>
>> ~~~ Andrew "lathama" Latham lath...@gmail.com ~~~
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] server performance....

[viswavardhanreddy karna]

HI,
 The way you said is correct, we are using SIPp to generate as many
calls as it can send and and the server is able is to take simultaneously of
560 - 570 calls

1. when we kept server for some time as idle it took 575 calls
2. when we kept again server as busy by continous calls back to back it is
taking 560-570 between i am not knowing which boundary should i take in
this

should i take the boundary of max successfull calls  when server is in busy
state or when server is in idle state?




Best Regards,
viswavardhan

On Fri, Mar 4, 2011 at 6:20 PM, Andrew Latham  wrote:

> On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
>  wrote:
> > Hi every one,
> >  I am doing some experiments on asterisk server
> > performance.. How can we know server performance? can any one explain
> me
> > plz
> >  I have 2 doubts regarding the asterisk server performance...
> >
> > 1. When can we know asterisk server performance?
> > 1. when server is in idle state ?
> > 2. when the server is in busy state?
> >
> > can any one please tell me when can the server performance is known i
> mean
> > when server is busy or in idle state?
> >
> > Best Regards,
> > viswavardhanreddy
>
>
> Many people test their servers with call-setups and call tear-downs.
> Using another tool like sipp you can send 100-1000s of call-setups and
> then do call tear-downs.  You can also use transcoding loops to test
> the load.  If you have 1 call that is sent to a context where it dials
> exten+1 and continues the loop until a target number, you can then set
> the codec for each dialed number.  I know that there are many methods
> of testing and this is just a common one.
>
> ~~~ Andrew "lathama" Latham lath...@gmail.com ~~~
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] server performance....

[Andrew Latham]

On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
 wrote:
> Hi every one,
>  I am doing some experiments on asterisk server
> performance.. How can we know server performance? can any one explain me
> plz
>  I have 2 doubts regarding the asterisk server performance...
>
> 1. When can we know asterisk server performance?
>     1. when server is in idle state ?
>     2. when the server is in busy state?
>
> can any one please tell me when can the server performance is known i mean
> when server is busy or in idle state?
>
> Best Regards,
> viswavardhanreddy


Many people test their servers with call-setups and call tear-downs.
Using another tool like sipp you can send 100-1000s of call-setups and
then do call tear-downs.  You can also use transcoding loops to test
the load.  If you have 1 call that is sent to a context where it dials
exten+1 and continues the loop until a target number, you can then set
the codec for each dialed number.  I know that there are many methods
of testing and this is just a common one.

~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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[asterisk-users] server performance....

[viswavardhanreddy karna]

Hi every one,
 I am doing some experiments on asterisk server
performance.. How can we know server performance? can any one explain me
plz
 I have 2 doubts regarding the asterisk server performance...


1. When can we know asterisk server performance?
1. when server is in idle state ?

2. when the server is in busy state?





can any one please tell me when can the server performance is known i mean
when server is busy or in idle state?








Best Regards,
viswavardhanreddy
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Re: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer

[vip killa]

I feel your pain

On Fri, Mar 4, 2011 at 9:29 AM, Danny Nicholas  wrote:

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis
> Carreiro
> Sent: Friday, March 04, 2011 8:07 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer
>
> Hey all,
>
> Alright. So we decided to not go with Avaya for our next PBX and we are now
> full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our
> SIP gateway and call center and Lync is our internal UC and IP-PBX server.
> I've already got Asterisk tied with our Nortel/Merridian Option 11 with
> QSig
> and all is beautiful (except for the Opt11 not receiving names from * but
> that's another topic). So, my problem now is with the call center.
>
> This setup may be a bit convoluted at first but it'll make sense I hope.
> I've created the queues in Asterisk via FreePBX. I then created a ring
> group
> for each Lync extension so we get the "Confirm Calls" option and dodge the
> voice mail problem. The agents the login via their Lync phone with the Ring
> Group extension as their Agent ID. It kind of looks like this:
>
> Queue 2001
>Agent 4001
>Agent 4002
>Agent 4003
>
> Ring Group 4001 -> Lync Extention 5001
> Ring Group 4002 -> Lync Extention 5002
> Ring Group 4003 -> Lync Extention 5003
>
> This all works beautifuly! The problem I have is on transfers. If Lync
> extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the
> transfer and shows that 5001 is still active with the call. We're using
> OrderlyStats to monitor the queue so I watch the "Talking" counter just
> keep
> counting instead of being aware the transfer took place. Now to me, that
> says to me that the transfer took place within Lync so Asterisk is unaware
> of the transfer. So my next step was to enable Refer support in Lync so
> Lync
> sends the refer message back to Asterisk to transfer the call so Asterisk
> is
> fully aware of what's going on. It seems like the refer message is trying
> to
> work and Lync is sending it and Asterisk is receiving it but the "Refer-To"
> is changing between the two so I'm at a loss.
>
> (Logs are below signature)
> Lync says it's sending the following message with a "Refer-to:
> "
>
> Asterisk is seeing the following and the refer-to changed, it's now
> "REFER-TO:
>
> 
> 7?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto
> -tag%3D8be38bb187>".
>
> At first it seems like Lync is sending a true SIP URI so I need to get
> Asterisk to know how to handle that SIP URI and then secondly, it seems
> like
> Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is
> this because Asterisk doesn't know how to handle the SIP URI?
>
> So I guess I'm left with wondering if fixing the REFER message stuff is
> going to fix my problem even? The end goal is for Asterisk to be aware that
> a call was transferred to another extension in Lync.
>
>
>
> Thanks in advance everyone!
> Louis
>
> 
>
> First of all, I assume you are using 1.8.X.  Regardless, Queueing and
> referring have some known issues.  If you look at chan_sip.c, you'll see
> that REFER is considered "broken" at this time (I know this to be the case
> in 1.4.37 and at least 1 flavor of 1.8).  So my suggestion is that you
> either devise some workaround for this or set up multiple queues so you can
> feed calls to these "phantom-busy" folks. My "Expertise" (such as it is) is
> at the AGI level; I only fool with the portions of the actual tree code
> that
> are patently obvious (usually tweaks to patches).
>
>
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Re: [asterisk-users] Testing from where number is...

[Benny Amorsen]

Piotr Górski  writes:

> So how to bill customers? Number portability makes it pretty impossible...

In the US, you pay the same to call a cell phone as you pay to call any
other phone. The callee pays for the airtime. This is a sensible
arrangement, as it allows for number portability and price competition.

Alas, Europe chose to pass the costs onto the caller, without even
making it reasonably possible for the caller to know whether he is
calling a cell phone or not! The Danish number plan in particular is
completely insane.


/Benny


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Re: [asterisk-users] Loudness of recorded wav-audio

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Friday, March 04, 2011 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Loudness of recorded wav-audio

Danny Nicholas wrote:
> In sip.conf, add rxgain=-4.0 to the peer.

The last I knew, rx/tx gains are only for dahdi/zaptel devices.

Doug

#1 You are probably correct
#2 As "copper usage" continues to drop and Asterisk progresses, I expect
this capability to be incorporated into the sip channel (if it isn't already
there in 1.8.X)


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Re: [asterisk-users] Loudness of recorded wav-audio

[Doug Lytle]

Danny Nicholas wrote:

In sip.conf, add rxgain=-4.0 to the peer.


The last I knew, rx/tx gains are only for dahdi/zaptel devices.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] Sangoma PCI vs PCI Express card

[Gopalakrishnan A.N]

PCI Express is always good also it take less power and faster when compare
to PCI in interupts .

On Fri, Mar 4, 2011 at 2:13 PM, Thorsten Göllner  wrote:

>  Am 03.03.2011 16:02, schrieb satish patel:
>
> Hey Guy,
>
> I have quick question. I am purchasing Sangoma A102D card but i am confused
> between PCI and PCI Express. Which card would be good for me.
>
> Definitely PCI Express is advance but i just want to know is there any
> major difference, like quality, performance etc..
>
>
> As far as I know you should prefer PCI Express. There should be less
> problems with IRQ-Sharing and IRQ-Overruns. We use a A104D (PCIe) and have
> no problems with the current driver set.
>
> --
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-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
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Re: [asterisk-users] Loudness of recorded wav-audio

[Danny Nicholas]

Defaults are 0.0 (leave volume unchanged)  +values make volume louder, -
softer.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Loudness of recorded wav-audio

 

Could yoz tell me the default value of rxgain or txgain, if there is no
rxgain or txgain in conf-data defined?

Von meinem iPad gesendet


Am 04.03.2011 um 15:34 schrieb "Danny Nicholas" :

In sip.conf, add rxgain=-4.0 to the peer.  This (feel free to correct)
should reduce the incoming volume by 4 decibels. You'll have to do a "sip
reload" for this to take effect.

 


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Loudness of recorded wav-audio

 

Thank you! How can I reduce the RXgain?


Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" < 
da...@debsinc.com>:


  _  


From:  
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To:  
 asterisk-users@lists.digium.com
Subject: [asterisk-users] Loudness of recorded wav-audio

 

Hello,

 

I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.

Thanks a lot.


best regards

Felix 

 

two options are:

1.  reduce RXgain - assuming your are using Record() command
2.  use sox to reduce the volume;  something like sox -v .8 file1.wav
file2.wav

 

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Re: [asterisk-users] Loudness of recorded wav-audio

[Felix Dong]

Could yoz tell me the default value of rxgain or txgain, if there is no rxgain 
or txgain in conf-data defined?

Von meinem iPad gesendet

Am 04.03.2011 um 15:34 schrieb "Danny Nicholas" :

> In sip.conf, add rxgain=-4.0 to the peer.  This (feel free to correct) should 
> reduce the incoming volume by 4 decibels. You’ll have to do a “sip reload” 
> for this to take effect.
>  
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
> Sent: Friday, March 04, 2011 8:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Loudness of recorded wav-audio
>  
> Thank you! How can I reduce the RXgain?
> 
> 
> Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" :
> 
>> 
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
>> Sent: Friday, March 04, 2011 2:31 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Loudness of recorded wav-audio
>>  
>> Hello,
>>  
>> I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it 
>> in wav-audio at the Asterisk server. I found the loudness level of the 
>> recorded audio was too high comparing with the orginal audio. How can I 
>> ajust it, so that there will be no amplifier used for recording.
>> Thanks a lot.
>> 
>> best regards
>> 
>> Felix 
>>  
>> two options are:
>> reduce RXgain – assuming your are using Record() command
>> use sox to reduce the volume;  something like sox –v .8 file1.wav file2.wav
>>  
> 
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 
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[asterisk-users] GXW4004 - lines get stuck

[Mike]

Hi,

 

I have an issue with a GWX4004 used a as a VoIP trunk to PSTN lines
converter.  In some instances, lines get stuck (both parties hang up, but
the GXW4004 status shows "off hook" for the lines). It stays like this until
reboot.

 

Is there a specific setting I should be looking for? I couldn't find
anything about that specifically.

 

Mike

 

 

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Re: [asterisk-users] Loudness of recorded wav-audio

[Danny Nicholas]

In sip.conf, add rxgain=-4.0 to the peer.  This (feel free to correct)
should reduce the incoming volume by 4 decibels. You'll have to do a "sip
reload" for this to take effect.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Loudness of recorded wav-audio

 

Thank you! How can I reduce the RXgain?


Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" :


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To:  
asterisk-users@lists.digium.com
Subject: [asterisk-users] Loudness of recorded wav-audio

 

Hello,

 

I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.

Thanks a lot.


best regards

Felix 

 

two options are:

1.  reduce RXgain - assuming your are using Record() command
2.  use sox to reduce the volume;  something like sox -v .8 file1.wav
file2.wav

 

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Re: [asterisk-users] Loudness of recorded wav-audio

[Felix Dong]

Thank you! How can I reduce the RXgain?


Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" :

> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
> Sent: Friday, March 04, 2011 2:31 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Loudness of recorded wav-audio
> 
>  
> 
> Hello,
> 
>  
> 
> I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it 
> in wav-audio at the Asterisk server. I found the loudness level of the 
> recorded audio was too high comparing with the orginal audio. How can I ajust 
> it, so that there will be no amplifier used for recording.
> 
> Thanks a lot.
> 
> 
> best regards
> 
> Felix 
> 
>  
> 
> two options are:
> 
> reduce RXgain – assuming your are using Record() command
> use sox to reduce the volume;  something like sox –v .8 file1.wav file2.wav
>  
> 
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro
Sent: Friday, March 04, 2011 8:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer

Hey all,

Alright. So we decided to not go with Avaya for our next PBX and we are now
full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our
SIP gateway and call center and Lync is our internal UC and IP-PBX server.
I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig
and all is beautiful (except for the Opt11 not receiving names from * but
that's another topic). So, my problem now is with the call center.

This setup may be a bit convoluted at first but it'll make sense I hope.
I've created the queues in Asterisk via FreePBX. I then created a ring group
for each Lync extension so we get the "Confirm Calls" option and dodge the
voice mail problem. The agents the login via their Lync phone with the Ring
Group extension as their Agent ID. It kind of looks like this:

Queue 2001
Agent 4001
Agent 4002
Agent 4003

Ring Group 4001 -> Lync Extention 5001
Ring Group 4002 -> Lync Extention 5002
Ring Group 4003 -> Lync Extention 5003

This all works beautifuly! The problem I have is on transfers. If Lync
extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the
transfer and shows that 5001 is still active with the call. We're using
OrderlyStats to monitor the queue so I watch the "Talking" counter just keep
counting instead of being aware the transfer took place. Now to me, that
says to me that the transfer took place within Lync so Asterisk is unaware
of the transfer. So my next step was to enable Refer support in Lync so Lync
sends the refer message back to Asterisk to transfer the call so Asterisk is
fully aware of what's going on. It seems like the refer message is trying to
work and Lync is sending it and Asterisk is receiving it but the "Refer-To"
is changing between the two so I'm at a loss.

(Logs are below signature)
Lync says it's sending the following message with a "Refer-to:
"

Asterisk is seeing the following and the refer-to changed, it's now
"REFER-TO:
".

At first it seems like Lync is sending a true SIP URI so I need to get
Asterisk to know how to handle that SIP URI and then secondly, it seems like
Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is
this because Asterisk doesn't know how to handle the SIP URI? 

So I guess I'm left with wondering if fixing the REFER message stuff is
going to fix my problem even? The end goal is for Asterisk to be aware that
a call was transferred to another extension in Lync.



Thanks in advance everyone!
Louis



First of all, I assume you are using 1.8.X.  Regardless, Queueing and
referring have some known issues.  If you look at chan_sip.c, you'll see
that REFER is considered "broken" at this time (I know this to be the case
in 1.4.37 and at least 1 flavor of 1.8).  So my suggestion is that you
either devise some workaround for this or set up multiple queues so you can
feed calls to these "phantom-busy" folks. My "Expertise" (such as it is) is
at the AGI level; I only fool with the portions of the actual tree code that
are patently obvious (usually tweaks to patches).


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Re: [asterisk-users] Loudness of recorded wav-audio

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Loudness of recorded wav-audio

 

Hello,

 

I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.

Thanks a lot.


best regards

Felix 

 

two options are:

1.  reduce RXgain - assuming your are using Record() command
2.  use sox to reduce the volume;  something like sox -v .8 file1.wav
file2.wav

 

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[asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer

[Louis Carreiro]

Hey all,

Alright. So we decided to not go with Avaya for our next PBX and we are now 
full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP 
gateway and call center and Lync is our internal UC and IP-PBX server. I've 
already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all 
is beautiful (except for the Opt11 not receiving names from * but that's 
another topic). So, my problem now is with the call center.

This setup may be a bit convoluted at first but it'll make sense I hope. I've 
created the queues in Asterisk via FreePBX. I then created a ring group for 
each Lync extension so we get the "Confirm Calls" option and dodge the voice 
mail problem. The agents the login via their Lync phone with the Ring Group 
extension as their Agent ID. It kind of looks like this:

Queue 2001
Agent 4001
Agent 4002
Agent 4003

Ring Group 4001 -> Lync Extention 5001
Ring Group 4002 -> Lync Extention 5002
Ring Group 4003 -> Lync Extention 5003

This all works beautifuly! The problem I have is on transfers. If Lync 
extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the 
transfer and shows that 5001 is still active with the call. We're using 
OrderlyStats to monitor the queue so I watch the "Talking" counter just keep 
counting instead of being aware the transfer took place. Now to me, that says 
to me that the transfer took place within Lync so Asterisk is unaware of the 
transfer. So my next step was to enable Refer support in Lync so Lync sends the 
refer message back to Asterisk to transfer the call so Asterisk is fully aware 
of what's going on. It seems like the refer message is trying to work and Lync 
is sending it and Asterisk is receiving it but the "Refer-To" is changing 
between the two so I'm at a loss.

(Logs are below signature)
Lync says it's sending the following message with a "Refer-to: 
"

Asterisk is seeing the following and the refer-to changed, it's now "REFER-TO: 
".

At first it seems like Lync is sending a true SIP URI so I need to get Asterisk 
to know how to handle that SIP URI and then secondly, it seems like Asterisk 
doesn't even receive the same REFER-TO message that Lync sent. Is this because 
Asterisk doesn't know how to handle the SIP URI? 

So I guess I'm left with wondering if fixing the REFER message stuff is going 
to fix my problem even? The end goal is for Asterisk to be aware that a call 
was transferred to another extension in Lync.



Thanks in advance everyone!
Louis


= Begin Lync SIP message 

TL_INFO(TF_PROTOCOL) [0]0B10.1E88::03/04/2011-13:21:17.501.0004fcd9 
(SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 215606761
Instance-Id: 00011F02
Direction: outgoing
Peer: lyncserver.internal.domain:5070
Message-Type: request
Start-Line: REFER 
sip:lyncserver.internal.domain:5070;grid=ed392a6bc0344a30b0841cd69be137ed 
SIP/2.0
From: "" 
;epid=e9688aa93e;tag=8be38bb187
To: 
;epid=B3E26C1E76;tag=9227b8a39d
CSeq: 2 REFER
Call-ID: aa6f8871-4151-4149-ad5a-29ab941bf4d0
Via: SIP/2.0/TLS 
20.20.20.20:54166;branch=z9hG4bKEB39D72C.F05E7E34CF9EF4FD;branched=FALSE
Max-Forwards: 69
Via: SIP/2.0/TLS 172.16.2.29:53851;ms-received-port=53851;ms-received-cid=400
User-Agent: CPE/4.0.7577.107 OCPhone/4.0.7577.107 (Microsoft Lync 2010 Phone 
Edition)
Supported: ms-dialog-route-set-update
Refer-to: 
Referred-By: 
;ms-referee-uri="sip:500;phone-context=enterpr...@domainname.com;user=phone";ms-identity="MIIBxwYJKoZIhvcNAQcCoIIBuDCCAbQCAQExDzANBgkqhkiG9w0BAQUFADALBgkqhkiG9w0BBwExggGPMIIBiwIBATBkMFYxFDASBgoJkiaJk/IsZAEZFgRSb290MRcwFQYKCZImiZPyLGQBGRYHTmV0d29yazElMCMGA1UEAxMcUGluY2ggQSBQZW5ueSBSb290IEF1dGhvcml0eQIKH1RMewAABjANBgkqhkiG9w0BAQUFADANBgkqhkiG9w0BAQEFAASCAQDCqFfl86RgR0gyLlbf6kEtUbFIWu2zCaFlqOoBWH/1ftxR80WwlsM142hImIGJhugOnXGIpCPMwC5OsPlD5Zk5+J371SSiz0dLhTjC1EzQzTMiIf9xbAd+ohi+e46TfXrJSN+esDNAafFKau0sj4hqPuuMh4EsabcNPrZwmroF5FX7zzEmRiavIKdGjyFgtwrwTIVkWjOnKdJdZN7T9jUj7Eb5HJGvOuG29LeJajTbogLugzsRA85RGQ7s+HMhU8F/RTJwQEvDhM4HQNeRFqUENi4jkYFUjuQaL3A2+Zqoml6LeZQNKU/afq8L9vck9+5qL1tc7c3+BtT+RxwlE92C:Fri,
 04 Mar 2011 13:21:17 
GMT";ms-identity-info="sip:Lyncserver.internal.domain:5061;transport=tls";ms-identity-alg=rsa-sha1
Content-Length: 0
P-Asserted-Identity: 
Privacy: id
Message-Body: -
$$end_record
= End Lync SIP message 



= Begin Asterisk Debug 

[Mar  4 08:21:05] DEBUG[18506] chan_sip.c:  Header  0 [ 53]: REFER 
sip:500@10.10.10.10:5067;transport=TLS SIP/2.0
[Mar  4 08:21:05] DEBUG[18506] chan_sip.c:  Header  1 [ 78]: FROM: 
;epid=431D53633D;tag=42b6d8c72b
[Mar  4 08:21:05] DEBUG[18506] chan_sip.c:  Header  2 [ 46]: TO: 
;tag=as0d823373
[Mar  4 08:21:05] DEBUG[18506] chan_sip.c:  Header  3 [ 13]: CSEQ: 2 REFER
[Mar  4 08:21:05

Re: [asterisk-users] Gosub and 'h' (again?)

[Andrew Thomas]

Nevermind - I've re-written my dialplan so that all subs are in one
context.  Now I only need 1 more line of code.

Thanks


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: 04 March 2011 11:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)


Problem as follows:

[default]
exten => 777,1,Gosub(sub,1,1)
exten => 777,n,Hangup()
exten => h,1,NoOp(hung up in 'default' context)

[sub]
exten => 1,1,NoOp(in sub)
exten => 1,n,Playback(tt-monkeys)
exten => 1,n,Return()
exten => h,1,NoOp(hung up in 'sub' context)

This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup.  You get the hang up in the 'default' context.

But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context.  This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =>
h,1,Goto(default,h,1)' in this case).

Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?

I know Tilghman said "...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place" [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].

But what does that mean in English ;)?

Thanks




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[asterisk-users] Gosub and 'h' (again?)

[Andrew Thomas]

Problem as follows:

[default]
exten => 777,1,Gosub(sub,1,1)
exten => 777,n,Hangup()
exten => h,1,NoOp(hung up in 'default' context)

[sub]
exten => 1,1,NoOp(in sub)
exten => 1,n,Playback(tt-monkeys)
exten => 1,n,Return()
exten => h,1,NoOp(hung up in 'sub' context)

This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup.  You get the hang up in the 'default' context.

But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context.  This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =>
h,1,Goto(default,h,1)' in this case).

Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?

I know Tilghman said "...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place" [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].

But what does that mean in English ;)?

Thanks




 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Testing from where number is...

[A J Stiles]

On Thursday 03 Mar 2011, Piotr Górski wrote:
> As free I mean no subscription. I can write AGI that will query
> numberingplans.com - that's not a problem... but I can query site only 20
> times a day without a subscription... So it's not free.

Well, free is as free does  :)

For the time being, keep making your 20 free queries per IP address per day, 
and build up a local MySQL database.  Populate it also from any other data 
sources you have available  (maybe you have letters with addresses and phone 
numbers? .)  Then have your AGI script always look in the local database 
first.  If what you need is not in there, and you still have some free 
queries remaining today  (even this information can be held within the 
database),  query numberingplans.com and save the result in your database.  
If you have run out of free queries, then you'll have to return something 
less precise  (just a country, perhaps; this information at least should be 
in your phone directory).

I can tell you now for free that 44 is the code for the UK; and UK numbers 
beginning with (0)7 are mobiles, (0)20 is London and (0)28 is Northern 
Ireland  :)

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] mySQL connection testing

[Tilghman Lesher]

On Friday 04 March 2011 03:03:41 Andrew Thomas wrote:
> Thanks Tilghman - this is exactly what I wanted to hear.  As for the
> 'inclusion' bit - true, but it's still infused in to the addons package
> at the Digium end (isn't it?).

While Digium hosts the repository and the project head (Russell) is a
Digium employee, what winds up in the repository is largely up to the
Asterisk community, including many non-Digium developers with commit
access.  While Digium does contribute a great deal to the releases,
suggesting that Digium is responsible for everything that ends up in a
release is reductionist and diminutive of the many contributions made by
the community.

-- 
Tilghman

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Re: [asterisk-users] mySQL connection testing

[Andrew Thomas]

Thanks Tilghman - this is exactly what I wanted to hear.  As for the
'inclusion' bit - true, but it's still infused in to the addons package
at the Digium end (isn't it?).

Anyway, I'll go create a mysql.conf file now :)

Cheers

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 04 March 2011 08:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing


On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote:
> Does anybody know of a way to test whether a mySQL connection invoked 
> from the dialplan is current or not?

There is no way to test it.  If you want this, you should track the
information yourself or don't disconnect anywhere but in the "h"
extension.

BTW, the disconnect is not strictly needed in all versions of the addons
since 1.4.9.  Due to the possibility of a memory leak, the connections
are tracked and deleted when the channel is destroyed.

See this issue (and the patch) for more information:
https://issues.asterisk.org/view.php?id=14757

-- 
Tilghman

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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Converting MP3 files to wav for Asterisk

[Ishfaq Malik]

On Thu, 2011-03-03 at 08:19 -0800, Steve Edwards wrote:
> Try something 'simpler'
> 
> mpg123 -q -w "${TEMP}" "${INPUT}"
> sox "${TEMP}" -c 1 -s -w -r 8000 "${OUTPUT}"
> 
> and see if that helps. Otherwise, how do the 'intermediate' files in
> your 
> process sound? Can you hear when things fall apart? 

I had been having the same issue and this above method has really
improved the quality of my wav files (I had previously been using
sox -V "${INPUT}" -r 8000 -c 1 -s "${OUTPUT}" resample -ql)

Thanks for that

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] mySQL connection testing

[Tilghman Lesher]

On Friday 04 March 2011 02:47:56 Andrew Thomas wrote:
> If mySQL in the dialplan is so bad - why did Digium include it
> in the first place?

Digium is not responsible for everything that appears in Asterisk.  This is
a community project, and community volunteers have written large swaths
of Asterisk, including the MYSQL command.

-- 
Tilghman

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Re: [asterisk-users] mySQL connection testing

[Tilghman Lesher]

On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote:
> Does anybody know of a way to test whether a mySQL connection invoked
> from the dialplan is current or not?

There is no way to test it.  If you want this, you should track the
information yourself or don't disconnect anywhere but in the "h"
extension.

BTW, the disconnect is not strictly needed in all versions of the addons
since 1.4.9.  Due to the possibility of a memory leak, the connections
are tracked and deleted when the channel is destroyed.

See this issue (and the patch) for more information:
https://issues.asterisk.org/view.php?id=14757

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Tilghman

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Re: [asterisk-users] mySQL connection testing

[Andrew Thomas]

Danny - Thanks, but that wouldn't work either - as I am fetching
multiple rows (not in that example - but I do in a production
environment).

Steve - If mySQL in the dialplan is so bad - why did Digium include it
in the first place?  JFYI - I use mySQL in the dialplan all the time -
and it always works a treat - first time, every time.  I do use AGI for
'other' things (eg. I've completely re-written the AgentCallbackLogin
feature in php) and that also works a treat. Each to their own I guess.

Anyway - back to the question (repeated in case it got lost amongst all
this) "Is there a way to check if a specific MYSQL connection id is
connected or not?".

BTW - using a 'disconnect {connid}' twice doesn't actually break
anything - it just causes an error on the console.  So I can live with a
'no' answer.

Thanks


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 03 March 2011 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing


On Thu, 3 Mar 2011, Andrew Thomas wrote:

> Gentlemen, can we please not turn this in to an Asterisk and DB 
> commands
> bashing thread?

I'm just suggesting that maybe you are 'swimming upstream' trying to use

MySQL within the dialplan.

Much the same as if you were proposing an office system using a 'tin
cans 
and string' mesh with carrier pigeons for out of band call signaling and

having a problem with poop buildup on the endpoints -- I might propose 
using Asterisk :)

-- 
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] Sangoma PCI vs PCI Express card

[Thorsten Göllner]

  
  
Am 03.03.2011 16:02, schrieb satish patel:

  
  Hey Guy,
  
  I have quick question. I am purchasing Sangoma A102D card but i am
  confused between PCI and PCI Express. Which card would be good for
  me. 
  
  Definitely PCI Express is advance but i just want to know is there
  any major difference, like quality, performance etc.. 


As far as I know you should prefer PCI Express. There should be less
problems with IRQ-Sharing and IRQ-Overruns. We use a A104D (PCIe)
and have no problems with the current driver set.
  


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[asterisk-users] Loudness of recorded wav-audio

[Felix Dong]

Hello,

I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.
Thanks a lot.

best regards

Felix
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