[asterisk-users] Confbridge examples for Asterisk 10?

[cjwstudios]

Does anyone have any application examples for Confbridge in Asterisk
10?  I'm just looking for simple ad-hoc functionality similar to
meetme in 1.8.  Thank you in advance.

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[Mitul Limbani]

Same for E1 as well unless your operator is giving mfcr2 on cas.

Mitul
On Jul 26, 2012 9:17 AM, "Jorge Mendoza"  wrote:

> Thank you Mitul for your answer.
> Yes, we have tested e&m before and it works. But I don't know why. That is
> what I don't understand. You said that E&M signalling does not have
> separate signalling channel, that is true for the T1 but not for E1. My
> understanding is that E1 CAS pass the E&M information in the bits abcd of
> channel 16, the signalling channel.
> Regards
> --
> Jorge Mendoza
>
> --
> *From: *"Mitul Limbani" 
> *To: *"Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Sent: *Wednesday, 25 July, 2012 8:15:25 PM
> *Subject: *Re: [asterisk-users] Dahdi+Redfone+Channel Bank+E&M
>
> E&M signalling do not have seperate signalling channel.
>
> Configure as e&m=1-31
>
> Mitul
> On Jul 26, 2012 6:40 AM, "Jorge Mendoza"  wrote:
>
>> Hi,
>>
>> We are trying to connect an Asterisk server with a Channel Bank with E&M
>> interfaces using a RedFone TDMoE device.
>> The CB have a E1 CAS interface.
>> OS: Ubuntu Server 11.10 64 bits
>> dahdi: dahdi-linux-complete-2.6.1+2.6.1
>>
>> Redfone configuration:
>> /etc/redfone.conf
>>
>> [span1]
>> framing=cas
>> encoding=hdb3
>>
>> System configuration:
>> /etc/dahdi/system.conf
>>
>> dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1
>> dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0
>>
>> e&me1=1-15,17-31
>> dchan=16
>>
>> alaw=1-31
>>
>> loadzone = fr
>> defaultzone = fr
>>
>> Error message:
>> 
>> # dahdi_cfg -v
>>
>> Changing signalling on channel 1 from Unused to E & M E1
>> Changing law on channel 1 from Mu-law to A-law
>> DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
>> Selected signaling not supported
>> Possible causes:
>> e&me1 signaling is being used on a T1 line (use e&m)
>> RBS signaling is being used on a E1 CCS span
>> Signaling is being assigned to channel 16 of an E1 CAS span
>> =
>>
>> I don't understand the last possible cause mentioned: "Signaling is being
>> assigned to channel 16 of an E1 CAS span", because the dchan is channel 16.
>> Where is the error?
>> Thank you.
>> --
>> Jorge Mendoza
>>
>> --
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>
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[Jorge Mendoza]

Thank you Mitul for your answer. 
Yes, we have tested e&m before and it works. But I don't know why. That is what 
I don't understand. You said that E&M signalling does not have separate 
signalling channel, that is true for the T1 but not for E1. My understanding is 
that E1 CAS pass the E&M information in the bits abcd of channel 16, the 
signalling channel. 
Regards 
-- 
Jorge Mendoza 

- Original Message -

From: "Mitul Limbani"  
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
 
Sent: Wednesday, 25 July, 2012 8:15:25 PM 
Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+E&M 


E&M signalling do not have seperate signalling channel. 
Configure as e&m=1-31 
Mitul 
On Jul 26, 2012 6:40 AM, "Jorge Mendoza" < jmendo...@tcc.com.pe > wrote: 


Hi, 

We are trying to connect an Asterisk server with a Channel Bank with E&M 
interfaces using a RedFone TDMoE device. 
The CB have a E1 CAS interface. 
OS: Ubuntu Server 11.10 64 bits 
dahdi: dahdi-linux-complete-2.6.1+2.6.1 

Redfone configuration: 
/etc/redfone.conf 

[span1] 
framing=cas 
encoding=hdb3 

System configuration: 
/etc/dahdi/system.conf 

dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1 
dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0 

e&me1=1-15,17-31 
dchan=16 

alaw=1-31 

loadzone = fr 
defaultzone = fr 

Error message: 
 
# dahdi_cfg -v 

Changing signalling on channel 1 from Unused to E & M E1 
Changing law on channel 1 from Mu-law to A-law 
DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) 
Selected signaling not supported 
Possible causes: 
e&me1 signaling is being used on a T1 line (use e&m) 
RBS signaling is being used on a E1 CCS span 
Signaling is being assigned to channel 16 of an E1 CAS span 
= 

I don't understand the last possible cause mentioned: "Signaling is being 
assigned to channel 16 of an E1 CAS span", because the dchan is channel 16. 
Where is the error? 
Thank you. 
-- 
Jorge Mendoza 

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[Mitul Limbani]

E&M signalling do not have seperate signalling channel.

Configure as e&m=1-31

Mitul
On Jul 26, 2012 6:40 AM, "Jorge Mendoza"  wrote:

> Hi,
>
> We are trying to connect an Asterisk server with a Channel Bank with E&M
> interfaces using a RedFone TDMoE device.
> The CB have a E1 CAS interface.
> OS: Ubuntu Server 11.10 64 bits
> dahdi: dahdi-linux-complete-2.6.1+2.6.1
>
> Redfone configuration:
> /etc/redfone.conf
>
> [span1]
> framing=cas
> encoding=hdb3
>
> System configuration:
> /etc/dahdi/system.conf
>
> dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1
> dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0
>
> e&me1=1-15,17-31
> dchan=16
>
> alaw=1-31
>
> loadzone = fr
> defaultzone = fr
>
> Error message:
> 
> # dahdi_cfg -v
>
> Changing signalling on channel 1 from Unused to E & M E1
> Changing law on channel 1 from Mu-law to A-law
> DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
> Selected signaling not supported
> Possible causes:
> e&me1 signaling is being used on a T1 line (use e&m)
> RBS signaling is being used on a E1 CCS span
> Signaling is being assigned to channel 16 of an E1 CAS span
> =
>
> I don't understand the last possible cause mentioned: "Signaling is being
> assigned to channel 16 of an E1 CAS span", because the dchan is channel 16.
> Where is the error?
> Thank you.
> --
> Jorge Mendoza
>
> --
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[Jorge Mendoza]

Hi,

We are trying to connect an Asterisk server with a Channel Bank with E&M 
interfaces using a RedFone TDMoE device.
The CB have a E1 CAS interface.
OS: Ubuntu Server 11.10 64 bits
dahdi: dahdi-linux-complete-2.6.1+2.6.1

Redfone configuration:
/etc/redfone.conf

[span1]
framing=cas
encoding=hdb3

System configuration:
/etc/dahdi/system.conf

dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1
dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0

e&me1=1-15,17-31
dchan=16

alaw=1-31

loadzone = fr
defaultzone = fr

Error message:

# dahdi_cfg -v

Changing signalling on channel 1 from Unused to E & M E1
Changing law on channel 1 from Mu-law to A-law
DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Selected signaling not supported
Possible causes:
e&me1 signaling is being used on a T1 line (use e&m)
RBS signaling is being used on a E1 CCS span
Signaling is being assigned to channel 16 of an E1 CAS span
=

I don't understand the last possible cause mentioned: "Signaling is being 
assigned to channel 16 of an E1 CAS span", because the dchan is channel 16.
Where is the error?
Thank you.
--
Jorge Mendoza

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[asterisk-users] SIP/GSM-gateway recommendation?

[Stefan Gofferje]

Hi,

can anybody recommend a priceworthy SIP/GSM-gateway that's known to work
flawlessly with asterisk?
Should especially support CLIP/CLIR in both directions and it would be
perfect if it would send notifications e.g. if the incoming call is
diverted or if the remote party puts me on hold.

I don't favor GSM-PCI-cards because I'm just building a new asterisk
based an an Atom board in a small casing.

--Stefan

-- 
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 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler & Koch - the original point and click interface


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Re: [asterisk-users] Finding the position of a character in a string

[Ishfaq Malik]

On Tue, 2012-07-24 at 12:45 +0200, giovanni.v wrote:
> Il 24/07/2012 10.37, Ishfaq Malik ha scritto:
> > It there a native asterisk dialplan function which will tell me the
> > position of a specific character in a given string?
> >
> > eg if I wanted to find what position the '@' was at in ${SIPURI}
> 
> if you are trying to extract parts from a string then look at the 
> function called 'CUT'.
> 
> verbose example:
> exten => s,1,NoOp(set DID from SIP TO header)
> exten => s,n,Set(DID_INFO=${SIP_HEADER(To)})
> exten => s,n,Set(DID_INFO=${CUT(DID_INFO,@,1)})
> exten => s,n,Set(DID_INFO=${CUT(DID_INFO,:,2)})
> exten => s,n,Goto(sip-routing,${DID_INFO},1)
> 
> 

Hi

That was exactly what I was after.

Thanks

Ish

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Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
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COMPANY REG NO. 04920552


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Re: [asterisk-users] Less good call quality using Asterisk

[Kevin P. Fleming]

On 07/25/2012 06:42 AM, Stefan at WPF wrote:

Hmm is it possible, that the monitor command changes the quality? If not
I guess I also once have to try compiling it from source, though I
wanted to avoid that.


It certainly can, since recording the call causes disk I/O as the audio 
is written out. In addition, Monitor is more prone to this problem than 
MixMonitor is, because Monitor's call recording is done in the same 
thread that handles the call's audio normally. If you switch to 
MixMonitor, you'll probably have better results, unless your system just 
can't handle recording the call without overloading its CPU.


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

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Re: [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?

[Tom Browning]

> You might want to check/compare disk-io & throughput on your G5 vs G7.
> Just a thought

Thanks Hans, I will do some disk benchmarking just in case.  I do know
that I/O wait on the G7s has been an order of magnitude less than the
G5s under the same load so I *think* the fancier raid device and
faster disks are doing their job.

Since I disabled Asterisk cdrs completely, the problem has gone away.
And I'm now on Asterisk 1.8.14.1 which appears to not crash like
1.8.12 did.

I'm going to try and recreate the problem without any of my AGI code
(just Asterisk extension B2BUA calling and CDRs enabled to default CSV
settings).

I'm guessing that perhaps there might be an Asterisk performance issue
writing to that flat file.

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[asterisk-users] How to play DTMF digits without blocking

[' Barros ']

Hi there,
I need a way to play digits received from user's phone without to block line 
from receive some more. For example:
exten => test,1,playback(type_your_age)exten => test,n,read(aux,,1)exten => 
test,n,saydigits(${aux}) ;HELP ME: In this moment, user can't type anything... 
I would like to play digit without force the user to pause.exten => goto(2)
There's no need to be with read application.
Thanks your help.
Reginaldo d'barros+5504185 9919 6279  --
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Re: [asterisk-users] res_odbc crashing asterisk after freetds dsn reconnects

[Matthew Jordan]

- Original Message - 

> From: "Noah Engelberth" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, July 25, 2012 8:10:04 AM
> Subject: [asterisk-users] res_odbc crashing asterisk after freetds
> dsn reconnects



> I do have a 91MB “core” file from yesterday’s incident in my /tmp
> directory (I assume that’s a core dump from the crash?)

Please use the instructions linked below to generate a backtrace
from a core dump.

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Note that you may need to re-compile Asterisk with the appropriate
build options to get a usable backtrace.

Once you have a backtrace, please file an issue on the issue tracker.

https://issues.asterisk.org/jira/

Thanks!

--
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[asterisk-users] res_odbc crashing asterisk after freetds dsn reconnects

[Noah Engelberth]

I have an Asterisk Open Source 10 system set up that is using res_odbc to 
connect to a MSSQL database so that our users can clock in/out on our timeclock 
system from their phones.  I've been having a consistent issue with Asterisk 
crashing (completely restarting and dropping active calls) when there is a 
network disruption that severs the connection between Asterisk and the MSSQL 
server while someone is trying to punch the timeclock.

The setup is as follows:
Asterisk 10.4 (also had same issues on 10.2) running on CentOS 6.2 (VM on a 
CentOS 6.3 KVM host cluster) - connected to Voice VLAN

-  freetds installed from epel yum repository, 0.91-2.el6 (most current 
version available on epel)

-  unixODBC & unixODBC-devel 2.2.14-11.el6 installed

-  Asterisk also has an ODBC connection to a local MySQL server 
configured and in use for a separate purpose
MSSQL 2008 R2 running on Server 2008 R2 (VM on a CentOS 6.3 KVM host cluster) - 
connected to Data VLAN

The res_odbc.conf file:
[my_freetds_dsn]
enabled => yes
dsn => my_freetds_dsn
username => my_freetds_user
password => my_freetds_password
pre-connect => yes
sanitysql => select 1

odbcinst.ini:

[FreeTDS]
Description = ODBC for Microsoft SQL
Driver= /usr/lib64/libtdsodbc.so.0
UsageCount   = 1
Threading= 2

odbc.ini:
[my_freetds_instance]
Description = my_freetds_instance
Driver = FreeTDS
Database = my_freetds_instanace
Server = my_mssql_server
Trace = no
TDS_Version = 7.2
Port = my_mssql_port
timeout = 10
connect_timeout = 5

The steps to replicate the crash are:

1)  Network disruption that prevents the Asterisk server from communicating 
with the MSSQL server occurs.

2)  While the network disruption is ongoing, a user dials into the Asterisk 
server's timeclock extension and inputs their employee ID, which causes 
Asterisk to perform a lookup on the MSSQL server.

3)  Asterisk "hangs" for 3-5 minutes while it waits for the ODBC connection 
to the MSSQL server.

4)  I get made aware of the problem and log in to Asterisk.

5)  I execute "module reload res_odbc.so" and Asterisk reconnects 
successfully to the ODBC connection and can process new calls to the timeclock.

6)  The "hung" calls continue to show in "core show channels" even after 
the user hangs up and tries again (for what it's worth users, typically create 
3-4 hung calls each before one or more of them let me know.  I've seen anywhere 
from 5-20 hung calls at the times I've logged in to try to reconnect the ODBC 
connection).

7)  Asterisk crashes during or shortly after the module reload.  Sometimes 
I've sent one or more "channel request hangup" commands from the Asterisk CLI 
for the hung calls.  Sometimes it crashes immediately on the module reload, 
sometimes it runs for a few minutes after the reload.  I don't think it's ever 
run more than 5 minutes after I reload the ODBC connections.

I do have a 91MB "core" file from yesterday's incident in my /tmp directory (I 
assume that's a core dump from the crash?)

Thank you,

Noah Engelberth
System Administration
MetaLINK Technologies

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Re: [asterisk-users] Less good call quality using Asterisk

[Stefan at WPF]

Hmm is it possible, that the monitor command changes the quality? If not I
guess I also once have to try compiling it from source, though I wanted to
avoid that.

2012/7/23 Bakko 

> Hello,
>
> I tried Asterisk Confbridge with raspberry pi without audio issue.
>
> Asterisk was compiled from sources.
>
> http://www.voztovoice.org/?q=**node/553
>
> Regards
>
>
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Re: [asterisk-users] Asterisk

[Mitul Limbani]

NO SIP / IAX Trunking allowed on IP.

Apart from that everything is legitimate.

We do provide Hosted Asterisk connected with PRI circuits, do contact
off-line for more info.

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422




On Wed, Jul 25, 2012 at 2:47 PM, Herve Prime  wrote:

> Is it legal to use Asterisk in India?
> We are planning to use Asterisk for Fax server and auto attendant.
>
> Thanks
>
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[asterisk-users] Asterisk

[Herve Prime]

Is it legal to use Asterisk in India?
We are planning to use Asterisk for Fax server and auto attendant.

Thanks
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