Re: [asterisk-users] IAX and Variables

2013-10-07 Thread Carlos Rojas 
I thunk so


Let me see
-Original Message-
From: Mikhail Lischuk 
Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 08 Oct 2013 01:08:22 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 
Subject: Re: [asterisk-users] IAX and Variables

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Re: [asterisk-users] IAX and Variables

2013-10-07 Thread Mikhail Lischuk 
 

Wont this work? 

exten =>
18,1,Set(CDR(accountcode)=${IAXVAR(ACCOUNTID)}) 

accountcode is not
read-only property so it should be writeable. 

Phibee Network Operation
Center писал 07.10.2013 21:05: 

> Hi
> 
> a new small question
;=)
> 
> We have two Asterisk, connected in IAX2.
> 
> On the first, in
dialplan, we have:
> exten =>
_XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
> we sent into the
IAXVAR "ACCOUNTID" the accountcode.
> 
> On the second, in dialplan, we
have:
> exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
> 
>
That's work, the second server get the variable.
> 
> I want now said at
the second server that accountcode = 
> ${IAXVAR(ACCOUNTID)},
> for use
this accoundcode in CDR. On second server, in cdr_mysql.conf i have:
>

> [columns]
> alias start => calldate
> alias end => callend
> alias
clid => clid
> alias src => src
> alias dst => dst
> alias dcontext =>
dcontext
> alias channel => channel
> alias dstchannel => dstchannel
>
alias lastapp => lastapp
> alias lastdata => lastdata
> alias duration
=> duration
> alias billsec => billsec
> alias disposition =>
disposition
> alias amaflags => amaflags
> alias accountcode =>
accountcode
> alias userfield => userfield
> alias uniqueid =>
uniqueid
> 
> But where i can put into the config that for this cdr
entry accountcode 
> = ${IAXVAR(ACCOUNTID)} ?
> 
> thanks for your
help
> 
> jerome

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With Best Regards
Mikhail Lischuk

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[asterisk-users] IAX and Variables

2013-10-07 Thread Phibee Network Operation Center 

Hi

a new small question ;=)

We have two Asterisk, connected in IAX2.

On the first, in dialplan, we have:
exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
we sent into the IAXVAR "ACCOUNTID" the accountcode.


On the second, in dialplan, we have:
exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})

That's work, the second server get the variable.




I want now said at the second server that accountcode = 
${IAXVAR(ACCOUNTID)},

for use this accoundcode in CDR. On second server, in cdr_mysql.conf i have:

[columns]
alias start => calldate
alias end => callend
alias clid => clid
alias src => src
alias dst => dst
alias dcontext => dcontext
alias channel => channel
alias dstchannel => dstchannel
alias lastapp => lastapp
alias lastdata => lastdata
alias duration => duration
alias billsec => billsec
alias disposition => disposition
alias amaflags => amaflags
alias accountcode => accountcode
alias userfield => userfield
alias uniqueid => uniqueid

But where i can put into the config that for this cdr entry accountcode 
= ${IAXVAR(ACCOUNTID)} ?



thanks for your help

jerome



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Re: [asterisk-users] Phones flashing but not ringing

2013-10-07 Thread Matt Hamilton 

> Have you tried restarting the phone instead of Asterisk? I don't think that 
> Asterisk sends 
> separate commands to the bell and to the call LED. Since the LED is flashing, 
> it is likely that 
> the "SIP INVITE" signal from Asterisk is ok. Also the ring tone normally does 
> not come from 
> Asterisk itself.
> 

We tried it and it doesn't help. It's not one phone, multiple phones do it at 
the same time. I think it's related to Asterisk SLA  - maybe device states get 
messed up.

Matt
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Re: [asterisk-users] ADSL and VPN router

2013-10-07 Thread vortex 

If you are looking for software i would go for pfsense.
if your are looking for hardware, i would go for mikrotik.



Hello;

I am looking for ADSL that supporting VPN so we can connect to it from 
our IPhone using the VPN to be able to register at the asterisk PBX. 
Any recommended one that is doing fine with voice? Also, does it 
support bandwidth priority or shaping for the protocols?


Regards
Bilal




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Re: [asterisk-users] Multiple resetcdr calls have no effect

2013-10-07 Thread Murthy Gandikota 
To answer my question, set unanswered=yes in cdr.conf

 

Source:

http://lists.digium.com/pipermail/asterisk-users/2009-December/241749.ht
ml

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Monday, October 07, 2013 7:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Multiple resetcdr calls have no effect

 

Hi All

 

Using Asterisk 11. My dial plan has the following context:

 

[sip-guest]

exten => _!.,1, Answer

exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}])

exten => _!.,n, resetcdr(w)

exten => _!.,n, resetcdr(w)

exten => _!.,n, set(DNIS=${EXTEN})

exten => _!.,n, resetcdr(w)

exten => _!.,n, hangup()

 

I expected at least 3 CDR's in MySQL. I see only one. 

Tried playing with cdr.conf settings but apparently they have no effect.

The cdr_mysql.conf is pretty generic.

 

*CLI> cdr show status

 

Call Detail Record (CDR) settings

--

Logging: Enabled

Mode: Batch

Log unanswered calls: No

Log congestion: No

 

* Batch Mode Settings

---

Safe shutdown: Enabled

Threading model: Scheduler only

Current batch size: 0 records

Maximum batch size: 3 records

Maximum batch time: 1 second

Next batch processing time: 0 seconds

 

* Registered Backends

---

mysql

Adaptive ODBC

csv

cdr-custom

 

I checked /var/log/asterisk/cdr-csv/Master.csv.

There is only one record for the call.

 

Expected the resetcdr to be synchronous.

 

Any help would be gratefully acknowledged.

 

Thanks

 

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[asterisk-users] Ring Busy ?

2013-10-07 Thread Phibee Network Operation Center 

Hi

I use asterisk with Realtime/Mysql.

I have put a Call-limit at 1, but if the SIP account receive two call in 
same time,

the second call don't ring busy.

Ayone know a solution  for the second call get a busy ring ?

Best Regards
Jerome



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[asterisk-users] AppKonference 2.4

2013-10-07 Thread Paul Albrecht 
Hi,

I have released AppKonference 2.4 today.

This release includes a frame cache for asterisk frames which should be more 
efficient than the asterisk frame cache is for this type of application. It's 
ulaw/alaw only for now. 

This release also added a speaker list which the conference thread uses to mix 
speaking members. This should be more efficient for large conferences when 
relatively few members are speaking.

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Re: [asterisk-users] Phones flashing but not ringing

2013-10-07 Thread jg 
Have you tried restarting the phone instead of Asterisk? I don't think that Asterisk sends 
separate commands to the bell and to the call LED. Since the LED is flashing, it is likely that 
the "SIP INVITE" signal from Asterisk is ok. Also the ring tone normally does not come from 
Asterisk itself.


jg

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Re: [asterisk-users] Dahdi not detecting hangup when analog forwarding

2013-10-07 Thread Olivier 
After posting her, I've found this thread
http://forums.asterisk.org/viewtopic.php?f=1&t=83088

In it I read:
" The normal PSTN behaviour is calling party clearing, with a timeout of 2
to 6 minutes if the called party is the only one that clears."

Can someone explain a bit further why "calling party clearing" keeps
"analog forwarding from working" ?
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[asterisk-users] Phones flashing but not ringing

2013-10-07 Thread Matt Hamilton 
We have been using Asterisk SLA for a while with Cisco SPA series phones. Once 
in a while the phones flash, but not ring when a call comes in. We can pick it 
up and talk to the caller even if that's the case. 

This is pretty random (might not happen for couple of weeks). The quick 
solution is to restart Asterisk which we are trying to avoid. What might cause 
this? 

Thanks,
Matt
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[asterisk-users] Dahdi not detecting hangup when analog forwarding

2013-10-07 Thread Olivier 
Hello,

I've got a test setup with 2 asterisk boxes:

Asterisk1 with:
asterisk 11.5.1
dahdi 2.7.0.1
Digium TDM400 with 2 FXO ports

Asterisk2 with:
asterisk 11.5.1
dahdi 2.7.0
Digium TDM400 with 2 FXS ports


Asterisk1 has the following AEL  Dialplan:

context remote {
s => {
Answer();
Dial(DAHDI/g1/7005);
};
};


When a call from Asterisk2 comes in, it is correctly entering the above
remote context and an extension on Asterisk2 receives an incoming call:
bothe Asterisk extension are talking to each other and everything run fine
(two ways audio) except that hangup is not detected by Asterisk1 (no matter
which Asterisk2 extension is hanging up):

- Asterisk2 shows non ongoing call,
- Asterisk1 show 2 living channels: I need to kill one of them to restore
expected status.

When I'm using another dialplan with which an incoming call is passed to a
local extension, hangup is correctly detected.


Am I trying to do something that can't be done (forwarding from one line to
another) ?
Any clue ?
Suggestions.

Regards
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[asterisk-users] Multiple resetcdr calls have no effect

2013-10-07 Thread Murthy Gandikota 
Hi All

 

Using Asterisk 11. My dial plan has the following context:

 

[sip-guest]

exten => _!.,1, Answer

exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}])

exten => _!.,n, resetcdr(w)

exten => _!.,n, resetcdr(w)

exten => _!.,n, set(DNIS=${EXTEN})

exten => _!.,n, resetcdr(w)

exten => _!.,n, hangup()

 

I expected at least 3 CDR's in MySQL. I see only one. 

Tried playing with cdr.conf settings but apparently they have no effect.

The cdr_mysql.conf is pretty generic.

 

*CLI> cdr show status

 

Call Detail Record (CDR) settings

--

Logging: Enabled

Mode: Batch

Log unanswered calls: No

Log congestion: No

 

* Batch Mode Settings

---

Safe shutdown: Enabled

Threading model: Scheduler only

Current batch size: 0 records

Maximum batch size: 3 records

Maximum batch time: 1 second

Next batch processing time: 0 seconds

 

* Registered Backends

---

mysql

Adaptive ODBC

csv

cdr-custom

 

I checked /var/log/asterisk/cdr-csv/Master.csv.

There is only one record for the call.

 

Expected the resetcdr to be synchronous.

 

Any help would be gratefully acknowledged.

 

Thanks

 

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Re: [asterisk-users] is g729 codec free? or under license???

2013-10-07 Thread John Novack 


Darryl Moore wrote:

Thank you Steve, and I read a bit more on the web on this subject
including your own well reasoned page at
http://www.soft-switch.org/patents/index.html

However, despite wide acceptance of the patentability of such codecs
(unfortunately), whether they are in fact software patents or not
appears to be a matter of opinion. The FSF and Fedora both refer to
codec patents as being software patents.

http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/
http://fedoraproject.org/wiki/Software_Patents

A quick google search of both terms will show that there are a great
many people who see codec patents as software patents, so I don't think
I am alone there.



Law is ALWAYS open to interpretation, so that is not surprising.
See if you can get any lawyer, and especially a patent attorney, to give you a 
definitive answer! You will not get one.
Seldom will you ever get an "eggspurt legal opinion" Any good lawyer will tell you 
"maybe", or if there is any doubt don't do it!
Law is not precisely measurable. No meter or O'scope to assist here.
Any A**hole can sue anyone for the filing fee, and the results are up to the 
opinion of a judge or jury.
The lawyers want it that way, so it isn't ever going to be any different.

John Novack

--

Dog is my Co-pilot


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[asterisk-users] Dahdi incoming call detection and hangup detection durations.

2013-10-07 Thread Olivier 
Hi,

I've set an Asterisk 11 box with a TDM400 board and Dahdi 2.7.0.1.

I've connected an FXS port to an FXO one and issued a couple of channel
originate command to measure the duration Asterisk/Dahdi needs to detect a
dahdi call is coming in.

Basically, using EPOCH variable, I'm reading a 2 or 3s duration with the
followinf AEL2 dialplan:

context remote {
s => {
if ("x${DB(Start/FXS1)}" != "x") {
Duration=$[${EPOCH} - ${DB(Start/FXS1)}];
Verbose(0,Duration is ${Duration});
}
Answer();
Wait(5);
HangUp();
};
};

context mylocal {

1 => {
DB(Start/FXS1)="${EPOCH}";
Dial(DAHDI/1);
HangUp();
};
};


How should I rate this 2s or 3s duration ?
Can I shorten this value ?
On the opposite, which settings would significantly increase this duration ?


With the same king of dialplan, I observed hangup needed 4 or 5s to
propagate from one port to the other. How should I rate this duration ?
Can I also shorten this value ?

Regards
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[asterisk-users] ADSL and VPN router

2013-10-07 Thread bilal ghayyad 
Hello;

I am looking for ADSL that supporting VPN so we can connect to it from our 
IPhone using the VPN to be able to register at the asterisk PBX. Any 
recommended one that is doing fine with voice? Also, does it support bandwidth 
priority or shaping for the protocols?

Regards
Bilal-- 
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