Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

2013-05-20 Thread Richard Mudgett 
> On 05/16/2013 10:07 AM, sean darcy wrote:
> > On 05/16/2013 09:41 AM, sean darcy wrote:
> >> I have a call on gv over motif. I try to bridge it to another call
> >> over
> >> motif, but a different gv account, and I get congestion.
> >>
> >> motif only handles one 1 channel at a time??
> >>
> >> sean
> >>
> >
> > More:
> >
> > Two different motif sections. Two different xmpp sections.
> > xmpp shows both connections.
> >
> > sean
> >
> 
> I've made some headway, but not enough.
> 
> I have a motif call coming in. That bridges to a motif call going
> out.
> 
> 
> > -- Executing [s@incoming:2] Wait("Motif/+1-2927", "1") in
> > new stack
> > -- Executing [s@incoming:3] Answer("Motif/+1-2927", "") in
> > new stack
> > -- Executing [s@incoming:4] Set("Motif/+1-2927",
> > 
> > "crazygooglecid=+1<>@voice.google.com/srvenc-nrqQ46ayZknr1yPFlefF9PJfrsgA2TBA")
> > in new stack
> > -- Executing [s@incoming:5] Set("Motif/+1-2927",
> > "stripcrazysuffix=+1<>") in new stack
> > -- Executing [s@incoming:6] Set("Motif/+1-2927",
> > "CALLERID(all)=+1<>") in new stack
> > -- Executing [s@incoming:7] Goto("Motif/+1-2927", "relay")
> > in new stack
> > -- Goto (incoming,s,14)
> > -- Executing [s@incoming:14] Dial("Motif/+1-2927",
> > "motif/51/+1@voice.google.com") in new stack
> > -- Called motif/51/+1@voice.google.com
> > -- Motif/+1@voice.google.com-d119 is proceeding passing it
> > to Motif/+1-2927
> > [May 20 12:17:25] NOTICE[4323][C-0167]: chan_motif.c:1636
> > jingle_indicate: Don't know how to indicate condition '15'
> >   == Spawn extension (incoming, s, 14) exited non-zero on
> >   'Motif/+1<>-2927'
> 
> I set xmpp debug on. As far as I can see the incoming call (xmpp port
> <45> ) terminates because of "success" just after the outgoing call (
> xmpp <51> ) is set up. The called number rings 3 times. If it's
> answered, it's dead.

I think you are required to send a DTMF 1 to the gv call after answering.

Richard

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Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

2013-05-20 Thread sean darcy 

On 05/16/2013 10:07 AM, sean darcy wrote:

On 05/16/2013 09:41 AM, sean darcy wrote:

I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.

motif only handles one 1 channel at a time??

sean



More:

Two different motif sections. Two different xmpp sections.
xmpp shows both connections.

sean



I've made some headway, but not enough.

I have a motif call coming in. That bridges to a motif call going out.



-- Executing [s@incoming:2] Wait("Motif/+1-2927", "1") in new stack
-- Executing [s@incoming:3] Answer("Motif/+1-2927", "") in new stack
-- Executing [s@incoming:4] Set("Motif/+1-2927", 
"crazygooglecid=+1<>@voice.google.com/srvenc-nrqQ46ayZknr1yPFlefF9PJfrsgA2TBA") in new stack
-- Executing [s@incoming:5] Set("Motif/+1-2927", 
"stripcrazysuffix=+1<>") in new stack
-- Executing [s@incoming:6] Set("Motif/+1-2927", "CALLERID(all)=+1<>") 
in new stack
-- Executing [s@incoming:7] Goto("Motif/+1-2927", "relay") in new stack
-- Goto (incoming,s,14)
-- Executing [s@incoming:14] Dial("Motif/+1-2927", 
"motif/51/+1@voice.google.com") in new stack
-- Called motif/51/+1@voice.google.com
-- Motif/+1@voice.google.com-d119 is proceeding passing it to 
Motif/+1-2927
[May 20 12:17:25] NOTICE[4323][C-0167]: chan_motif.c:1636 jingle_indicate: 
Don't know how to indicate condition '15'
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1<>-2927'


I set xmpp debug on. As far as I can see the incoming call (xmpp port 
<45> ) terminates because of "success" just after the outgoing call ( 
xmpp <51> ) is set up. The called number rings 3 times. If it's 
answered, it's dead.




-- Called motif/<51>/+1@voice.google.com

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq from="+1@voice.google.com" to="<51>@gmail.com/asterisk-xFF200C8A" type="error" id="m">http://www.google.com/session";>http://www.google.com/session/phone";>xmpp:+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==http://www.google.com/session";>xmpp:+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==
<->

<--- XMPP sent to '<51>' --->

<->

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq to="<51>@gmail.com/asterisk-xFF200C8A" 
from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" id="n" type="result"/>
<->

<--- XMPP sent to '<51>' --->


<->
-- Motif/+1@voice.google.com-7abc is proceeding passing it to 
Motif/+1-5e4e
[May 20 12:19:53] NOTICE[4326][C-0168]: chan_motif.c:1636 jingle_indicate: 
Don't know how to indicate condition '15'

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq to="<51>@gmail.com/asterisk-xFF200C8A" 
from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" id="o" type="result"/>
<->

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" to="<51>@gmail.com/asterisk-xFF200C8A" id="jingle:10.13.61.5-23164143:1:AB47DEE4" type="set">http://www.google.com/session";>
<->

<--- XMPP sent to '<51>' --->

<->

<--- XMPP received from '<45>' --->
<
<->

<--- XMPP received from '<45>' --->
iq from="+1@voice.google.com/srvenc-xPt/DDObueIws5jdu1V1AmeBB2R4pgkM" to="<45>@gmail.com/asterisk-xAF820CA3" id="jingle:10.68.166.37-7896298:1:898DBA27" type="set">http://www.google.com/session";>Call endedhttp://www.google.com/session/phone"/>
<->

<--- XMPP sent to '<45>' --->

<->

<--- XMPP sent to '<51>' --->

<->
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1-5e4e'



The incoming motif also drops if the bridged call is SIP:


-- Called SIP//

<--- XMPP received from '<45>' --->
<
<->

<--- XMPP received from '<45>' --->
iq from="+1@voice.google.com/srvenc-jgCvKGNmqXXjKPKzhCuU3KoKurQ4+Fi1" to="<45>@gmail.com/asterisk-xAF820CA3" id="jingle:10.229.151.16-7608331:1:9E0372A9" type="set">http://www.google.com/session";>Call endedhttp://www.google.com/session/phone"/>
<->

<--- XMPP sent to '<45>' --->

<->
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1



Very odd.

Any help appreciated.

sean





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[asterisk-users] Stress testing Asterisk

2013-05-20 Thread Tommy Cooper 
Hi,
I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating 
are failing. I am trying to run Sipp on the same machine as Asterisk PBX using 
the ./sipp -sn uac 192.168.1.115 command.

SIpp output:
- Statistics Screen --- [1-9]: Change Screen --
  Start Time | 2013-05-20 22:53:08:637 1369083188.637273    
  Last Reset Time    | 2013-05-20 22:55:17:676 1369083317.676598    
  Current Time   | 2013-05-20 22:55:17:676 1369083317.676651    
-+---+--
  Counter Name   | Periodic value    | Cumulative value
-+---+--
  Elapsed Time   | 00:00:00:000  | 00:02:09:039 
  Call Rate  |    0.000 cps  |    0.930 cps 
-+---+--
  Incoming call created  |    0  |    0 
  OutGoing call created  |    0  |  120 
  Total Call created |   |  120 
  Current Call   |    0  |  
-+---+--
  Successful call    |    0  |    0 
  Failed call    |    0  |  120 
-+---+--
  Response Time 1    | 00:00:00:000  | 00:00:00:000 
  Call Length    | 00:00:00:000  | 00:00:31:509 
-- Test Terminated 
2013-05-20 22:55:17:675 1369083317.675242: Aborting call on UDP retransmission 
timeout for Call-ID '120-60749@192.168.1.114'.
sipp: There were more errors, enable -trace_err to log them.

This an error message I get when I use -trace_err:
2013-05-20 23:00:59:021    1369083659.021771: Aborting call on UDP 
retransmission timeout for Call-ID '33-60833@192.168.1.114


Thanks in advance.

Regards,
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Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen

2013-05-20 Thread Hans Witvliet 
-Original Message-
From: Rafael dos Santos Saraiva 
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] Performance Asterisk large installation on
Vmware/Xen
Date: Sat, 18 May 2013 15:01:06 -0300

Hi


I would like the opinion of you and if anyone has a similar scenario. I
have a project for installation of a Asterisk server in a client with
about 400 extensions. My question is whether this scenario carry an
Asterisk virtualized. Will be used only extensions and trunks sip sip, 1
queue with 2 agents, without call recording. It is best to use XEN or
VMware? Which best version of Asterisk for this scenario?
_


Use XEN in paravirtualized mode: NOT hardware/full virtualized!

Even when using specialized drivers, you get a considerable performance
hit. When virtualizing Linux, hw-virtualization is an unneeded waste of
cpu-cycles. Acceptable for windows clients, not otherwise.



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Re: [asterisk-users] How to allow AMI access to Originate yet deny Application: System

2013-05-20 Thread Alex Villací­s Lasso 

El 15/05/13 10:10, Alex Villací­s Lasso escribió:
While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, 
combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug in our system. However, there are third parties that plug into our system, and that reconfigure the 
manager.conf file to allow remote access to AMI logins that allow Originate (by default, the manager.conf remains configured to deny login to any system except localhost). I want to have a guideline on how to proceed in order to make these applications 
work, without allowing malicious users to compromise the system. I know that one way to proceed is to deny remote access to AMI, and build an application-specific proxy that will perform the Originate on behalf of the remote requester, after filtering 
the values. However, I want to know if there is a simpler way to remove the danger of code execution while allowing applications to use AMI to place calls.


The intended scenario is that a remote desktop application (for Windows) is configured with the AMI credentials, and connects over the LAN to Asterisk in order to place calls and otherwise monitor the system. The attack I want to protect against is that 
of a malicious user that collects the credentials from the desktop application and proceeds to use the Application: System trick. I know of the SSL support for AMI, but it will not protect against a malicious end user.

Is this question too basic, or just cannot be done other than with the proxy 
approach?

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Re: [asterisk-users] Passcode

2013-05-20 Thread sbasurto 
Hello Felix,

I did it in this way:

In the extensions.conf
Create this two Macros:

[macro-ask-pass]
exten => s,1,Set(ORI=${MACRO_EXTEN})
same => n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5
seconds
same => n,Set(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
same => n,Background(vm-password)
same => n,WaitExten(5)

[macro-verify-pass]
exten => s,1,Set(GO=${ODBC_PASS(${MACRO_EXTEN:2},${ARG1},${ARG2})})
same => n,Verbose(${GO})
same => n,Verbose(${ORI})
same => n,GotoIf($[${GO} = 0]?wrong:ok)
same => n(wrong),Playback(vm-incorrect)
same => n,Hangup()
same => n(ok),Playback(auth-thankyou)

Then in trunk ld I call the macro this way (all the passwords start with
*1 (for example *1252525252, *1545288362): 

[trunkld]

exten => _9X1NX,1,Macro(ask-pass)
exten => _*1.,1,Macro(verify-pass,restricted,unlimited)
same => n,Set(FILENAME=g
${CALLERID(all)}-${EXTEN}-${STRFTIME(${EPOCH},GMT+5,%C%y%m%d%H%
M)}-${UNIQUEID})
same => n,Macro(dundi-e164,${ORI:1})
same => n,MixMonitor(${MONDIR}${FILENAME}.wav,b)
same => n,Dial(${GLOBAL(TRUNK)}/${ORI:${GLOBAL(TRUNKMSD)}})
same => n,StopMixMonitor()

In the func_odbc.conf
[PASS]
dsn=asterisk
readsql=select count(*) from ast_passwd where passwd =
'${SQL_ESC(${ARG1})}' and user_profile in
('${SQL_ESC(${ARG2})}','${SQL_ESC(${ARG3})}')


I do not know if this is the better way to achieve this but this works,
and ask password every time some one call to long distance number.


I hope this helps.

Regards,
Sergio Basurto

On Mon, 2013-05-20 at 13:02 +, Felix Vazquez wrote:
> How do I make a user dial a passcode if he wants to make an
> international call?
> 
>  
> 
> 
> 
> 
> 
> __
> 
> This electronic message contains information from BOSH Global Services
> which may be company sensitive, proprietary, privileged or otherwise
> protected from disclosure. The information is intended to be used
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Re: [asterisk-users] Passcode

2013-05-20 Thread Leandro Dardini 
Again, the authenticate function can help you

Leandro


2013/5/20 Felix Vazquez 

>  How do I make a user dial a passcode if he wants to make an
> international call?
>
>
>
> --
>
> This electronic message contains information from BOSH Global Services
> which may be company sensitive, proprietary, privileged or otherwise
> protected from disclosure. The information is intended to be used solely by
> the recipient(s) named above. If you are not an intended recipient, be
> aware that any review, disclosure, copying, distribution or use of this
> transmission or its contents is prohibited. If you have received this
> transmission in error, please notify the sender immediately.
>
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Re: [asterisk-users] Secure Calling

2013-05-20 Thread Leandro Dardini 
I think it can be worth checking the authenticate function.

http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate


2013/5/20 Felix Vazquez 

>  How do I make a user dial a passcode to make  calls through asterisk?
>
> We would like to place a phone at a client’s location for our employee but
> are afraid it may get abused by the other workers.
>
>
>
> --
>
> This electronic message contains information from BOSH Global Services
> which may be company sensitive, proprietary, privileged or otherwise
> protected from disclosure. The information is intended to be used solely by
> the recipient(s) named above. If you are not an intended recipient, be
> aware that any review, disclosure, copying, distribution or use of this
> transmission or its contents is prohibited. If you have received this
> transmission in error, please notify the sender immediately.
>
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Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen

2013-05-20 Thread Angelo Delphini 
Rafael nice day.

Regarding the issue of virtualization I do not recommend running asterisk
in virtualized system.

As for the scenery, I now have this same scenario (260 extensions) and is
working perfectly ... Of course all account eg My server is a Dell
PowerEdge R310 and appliances are Grandstream GXP1405.

The network is all Gigabit.

As for the version I'm using 1.8.x LTS (
http://downloads.asterisk.org/pub/telephony/certified-asterisk/certified-asterisk-1.8.15-current.tar.gz
)

hugs
===
Angelo Delphini
Telecommunications Analyst
"Desempenho e Segurança através de Soluções em Dados, Voz e Vídeo"
E-mail: adelph...@camtecnologia.com.br
Gtalk: adelph...@camtecnologia.com.br
Skype: angelo.delphini
T: 55 21 2260-0324
T: 55 11 4063-0986
T: 55 21 9541-4757
Telefone VoIP: 55 21 1701-3000
CAM Tecnologia Ltda
www.camtecnologia.com.br

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2013/5/18 Rafael dos Santos Saraiva 

> Hi
>
> I would like the opinion of you and if anyone has a similar scenario. I
> have a project for installation of a Asterisk server in a client with about
> 400 extensions. My question is whether this scenario carry an Asterisk
> virtualized. Will be used only extensions and trunks sip sip, 1 queue with
> 2 agents, without call recording. It is best to use XEN or VMware? Which
> best version of Asterisk for this scenario?
>
> Thank you.
>
> Att,
> *Rafael dos Santos Saraiva*
> Tel: (51) 8174-7956 | (51) 3205-1504
> http://www.astdocs.com | 
> 
>
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Re: [asterisk-users] Loopback question

2013-05-20 Thread Leandro Dardini 
Is the "echo" application suitable to you?

Leandro


2013/5/20 CDR 

> Dear friends
> I need to loopback the audio on my channel. Did anybody on the development
> team thought about a function or app that would do that? If it is not
> clear, I mean that whatever audio I get, I send back.
> Philip
>
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Re: [asterisk-users] Question

2013-05-20 Thread Joshua Colp 

CDR wrote:

Is it me or Google just blocked Asterisk's chan_motif? I get "violation
of terms of service" audio message whenever I send a call.


Works fine here. Their automated security system probably determined 
your usage behavior was not consistent with normal usage and terminated 
your access.


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Check us out at:  www.digium.com  & www.asterisk.org

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[asterisk-users] Passcode

2013-05-20 Thread Felix Vazquez 
How do I make a user dial a passcode if he wants to make an international call?




This electronic message contains information from BOSH Global Services which 
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[asterisk-users] Secure Calling

2013-05-20 Thread Felix Vazquez 
How do I make a user dial a passcode to make  calls through asterisk?
We would like to place a phone at a client's location for our employee but are 
afraid it may get abused by the other workers.




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disclosure, copying, distribution or use of this transmission or its contents 
is prohibited. If you have received this transmission in error, please notify 
the sender immediately.
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[asterisk-users] Loopback question

2013-05-20 Thread CDR 
Dear friends
I need to loopback the audio on my channel. Did anybody on the development
team thought about a function or app that would do that? If it is not
clear, I mean that whatever audio I get, I send back.
Philip
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[asterisk-users] Question

2013-05-20 Thread CDR 
Is it me or Google just blocked Asterisk's chan_motif? I get "violation of
terms of service" audio message whenever I send a call.
Philip
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