Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

[Victor Villarreal]

Ok..

Maybe you can try with this command, from the Asterisk cli:

cli> core set verbose 5
cli> core set debug 5
cli> iax2 set debug on

And then, try to register with your softphone.

At the end of the test, execute:

cli> iax2 set debug off

And finally, review the logfiles.

2017-06-05 17:40 GMT-03:00 <the...@sys-concept.com>:

> Doesn't matter how much I increase the verbose output
> asterisk -vvr
> asterisk will not even print a single line.
>
> How to find out if my firewall has this port open?
> https://www.grc.com
> is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/
>
>
> Thelma
> On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> > I think you need to increase verbose output and search in
> > /var/log/asterisk/full for any error message related to IAX2 registration
> > or simil.
> >
> > 2017-06-05 17:12 GMT-03:00 <the...@sys-concept.com>:
> >
> >> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> >> while and it was zoiper was working OK with my previous version of
> >> asterisk.
> >>
> >> After upgrade to 11.25.1 it stop working.
> >> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> >> 443 open.
> >>
> >>
> >> Thelma
> >> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> >>> Another might be to make sure iptables isn't blocking the connection.
> >>>
> >>> You can run
> >>> iptables -L -n -v
> >>> To see if its set to block any ports.
> >>>
> >>>
> >>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
> >>>> I'm getting:
> >>>> netstat -a |grep 4569
> >>>> udp0  0 0.0.0.0:45690.0.0.0:*
> >>>>
> >>>> Should I be getting localhost IP?
> >>>>
> >>>> Thelma
> >>>>
> >>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> >>>>> Does asterisk listen on port 4569 by default?
> >>>>>
> >>>>> I'm running version Asterisk 11.25.1 and have a problem registering
> >>>>> Zoiper (IAX) to Asterisk.
> >>>>> I'm getting an error:
> >>>>> Registration refused
> >>>>>
> >>>>
> >>>> --
> >>>> _
> >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>>>
> >>>> Check out the new Asterisk community forum at:
> >>>> https://community.asterisk.org/
> >>>>
> >>>> New to Asterisk? Start here:
> >>>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>>>
> >>>> asterisk-users mailing list
> >>>> To UNSUBSCRIBE or update options visit:
> >>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >>>
> >>>
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> Check out the new Asterisk community forum at:
> https://community.asterisk.
> >> org/
> >>
> >> New to Asterisk? Start here:
> >>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> >
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

I think you need to increase verbose output and search in
/var/log/asterisk/full for any error message related to IAX2 registration
or simil.

2017-06-05 17:12 GMT-03:00 :

> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> while and it was zoiper was working OK with my previous version of
> asterisk.
>
> After upgrade to 11.25.1 it stop working.
> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> 443 open.
>
>
> Thelma
> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> > Another might be to make sure iptables isn't blocking the connection.
> >
> > You can run
> > iptables -L -n -v
> > To see if its set to block any ports.
> >
> >
> > On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
> >> I'm getting:
> >> netstat -a |grep 4569
> >> udp0  0 0.0.0.0:45690.0.0.0:*
> >>
> >> Should I be getting localhost IP?
> >>
> >> Thelma
> >>
> >> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> >>> Does asterisk listen on port 4569 by default?
> >>>
> >>> I'm running version Asterisk 11.25.1 and have a problem registering
> >>> Zoiper (IAX) to Asterisk.
> >>> I'm getting an error:
> >>> Registration refused
> >>>
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> Check out the new Asterisk community forum at:
> >> https://community.asterisk.org/
> >>
> >> New to Asterisk? Start here:
> >>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

No. The 0.0.0.0 listen address is fine.

El 5 jun. 2017 10:06,  escribió:

> I'm getting:
> netstat -a |grep 4569
> udp0  0 0.0.0.0:45690.0.0.0:*
>
> Should I be getting localhost IP?
>
> Thelma
>
> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> > Does asterisk listen on port 4569 by default?
> >
> > I'm running version Asterisk 11.25.1 and have a problem registering
> > Zoiper (IAX) to Asterisk.
> > I'm getting an error:
> > Registration refused
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

Another idea:

* Run netstat -tulpn command on Linux box AND look if there are an Asterisk
process listening on 4569 UDP port on 0.0.0.0

El 5 jun. 2017 10:00, "Victor Villarreal" <mefhigos...@gmail.com> escribió:

> Dear Thelma,
>
> Yes. Asterisk listen on port 4569 UDP on default config.
>
> Please, look at the Asterisk logfile, for clues about your issue. Or
> enable IAX2 debug vía Asterisk CLI.
>
> Other ideas:
>
> * Check that your server firewall permit UDP port 4569 incoming traffic.
>
> * Run tcpdump over the network interface of your server where the
> registration packets suppose come in. Look ir at least the softphone
> registration request are reaching the server.
>
> * Check if the credentials configured un the softphone mach the
> credentials configured on the server.
>
> Cheers
>
> El 5 jun. 2017 9:48, <the...@sys-concept.com> escribió:
>
>> Does asterisk listen on port 4569 by default?
>>
>> I'm running version Asterisk 11.25.1 and have a problem registering
>> Zoiper (IAX) to Asterisk.
>> I'm getting an error:
>> Registration refused
>>
>> --
>> Thelma
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX port 4569

[Victor Villarreal]

Dear Thelma,

Yes. Asterisk listen on port 4569 UDP on default config.

Please, look at the Asterisk logfile, for clues about your issue. Or enable
IAX2 debug vía Asterisk CLI.

Other ideas:

* Check that your server firewall permit UDP port 4569 incoming traffic.

* Run tcpdump over the network interface of your server where the
registration packets suppose come in. Look ir at least the softphone
registration request are reaching the server.

* Check if the credentials configured un the softphone mach the credentials
configured on the server.

Cheers

El 5 jun. 2017 9:48,  escribió:

> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>
> --
> Thelma
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Automatically dial a number, then an extension

[Victor Villarreal]

Hi John,

I think we need to known how you play the audio to the customers, before we
can help you.

Are you using AMI? Or AGI maybe? Or Call files?

What Asterisk version do you have?

El 15 may. 2017 12:35, "Tech Support"  escribió:

> All;
>
> I have an application that dials a list of numbers and then plays a
> recorded message. My customer uses it to dial a list of customers to
> confirm their appointment for the next day. No biggie, maybe 25 – 30 calls
> per day for customers who want the confirmation call. What they need now is
> a way to dial an extension after the number is dialed and answered. I’ve
> seen that before, but I just can't remember where. I was wondering if
> anyone else has implemented something along these lines. Any insight at all
> would be greatly appreciated.
>
> Thanks Much;
>
> John V.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

[Victor Villarreal]

Hi David, Tim,

Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that
permit the traffinc AND ONLY BLOCK them after certain level triggered.


Use iptables to block the unused services faced to public networks like
Internet. And configure these services properly, so they listen only
selected interfaces and IPs, and not from 0.0.0.0

2017-04-21 13:47 GMT-03:00 Tim S :

> Is that IP in your network or outside (I can ping it so I'm guessing it's
> outside your network)?  Do you have a firewall between your asterisk box
> and the internet?  Is there a WHITELIST of IP addresses that only allow
> your provider's limited IP pool to connect to your asterisk box from
> outside?
>
> If you are getting TFTP requests hitting your Asterisk box, they are not
> properly being filtered at your firewall - ftp and tftp are considered
> insecure communication methods, that port (69 I think) should be closed on
> your firewall unless you have a really good reason to have it opened (and
> unless you run a public FTP site, THERE IS NO GOOD REASON).
>
> Fail2Ban is a BLACKLIST method, blacklists are most effective after good
> network hygiene is implemented, as you drastically limit the pool of
> potential bad actors with a whitelist.
>
> Best,
>
> -Tim
>
> On Fri, Apr 21, 2017 at 9:38 AM, Dovid Bender  wrote:
>
>> This is old news. They use Shodan and then try to connect. Set up
>> Fail2Ban that say after 10 404's to ban the IP.
>>
>>
>> On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis 
>> wrote:
>>
>>> I "justed" happened to look at /var/log/messages...
>>>
>>> I saw:
>>> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6e.cfg
>>>
>>> so basically an sequential read of polycom MAC address config files.
>>> Some is trying to read to determine if I have any polycom files just
>>> sequential read after read.
>>> And if so - it would get any extension and password at that time.
>>> Luckily I have none.
>>>
>>> However - how does one block attempts like this ?
>>>
>>> Thanks!
>>>
>>> Jerry
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

[Victor Villarreal]

Hi, Jerry,

I don't know what S.O. you have in the Server, but you can check the man
page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options
--address, so you can tell tftp from what interface/port this service
listen request.

>From the IP in your logs (69.64.57.18) the request came from a web hosting
provider (http://www.heg.com/). So, the request came from Internet, so your
server listen TFTP request from outside, what is bad.

You can use iptables in any Linux distro to block incoming TFTP traffic.
TFTP is a UDP protocol at port 69.

Example:

/sbin/iptables -A INPUT -i eth0 -p udp --destination-port 69 -j DROP

Change eth0 to the correct name of your public internet server interface.



2017-04-21 13:27 GMT-03:00 Jerry Geis :

> I "justed" happened to look at /var/log/messages...
>
> I saw:
> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
> 0004f2034f6e.cfg
>
> so basically an sequential read of polycom MAC address config files.
> Some is trying to read to determine if I have any polycom files just
> sequential read after read.
> And if so - it would get any extension and password at that time.
> Luckily I have none.
>
> However - how does one block attempts like this ?
>
> Thanks!
>
> Jerry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Victor Villarreal]

Hi Ernie,

When one-way audio appear (no matters if there is a VPN or NAT server on
the diagram) I simply :

* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x'
on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want
to debug.

* Make a test call and replicate the issue.

* Stop debug with 'sip set debug off'.

* Follow the SIP conversation. Verify that the INVITE message has the
correct IP on the contact field and any other related fields.

* On SDP handshake, verify that the ports where the sound is send, is
correct.

Normally, one-way audio is faced when one audio stream (example the called
audio) is send to the correct IP and Port destination, on the other audio
stream (example the caller audio) don't.

Last, if Asterisk is 'behind' another server, you need tell Asterisk what
is the external IP so it can inform this IP to your clients.

If you dont want to follow the SIP conversation on plain text, you can make
a packet capture on the Asterisk server, instead of SIP debug.

El 19 abr. 2017 16:38, "Mark Wiater"  escribió:

> On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
>
>> Server network: 192.168.0.0/24
>> OpenVPN network: 10.8.0.0/24
>> Asus network: 192.168.1.0/24
>>
>> The Asterisk SIP registration appears to be responding properly to this -
>> this is what I see when I do a 'sip show peer' for an Aastra phone that's
>> connecting through the VPN (Asterisk output is truncated):
>>
>>   ToHost   :
>>   Addr->IP : 10.8.0.6:5060
>>
>
> If the Asus network is 192.168.1.0/24, and the phone is registering as
> 10.0.8.6, it looks like NAT is taking place. Would your asterisk server
> know how to route traffic to 192.168.1.0/24?
>
> I've always used site-to-site OpenVPN tunnels where the vpn's terminate on
> the gateway for both the phones and the asterisk server. I've always had
> rock solid connections between phones and Asterisk.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voicemail asking for login

[Victor Villarreal]

Hi Darcy,

What Pete think is correct.

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.

Finally, in the Asterisk wiki you can find more info:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes

Cheers

El 18 abr. 2017 21:18, "Pete Mundy"  escribió:

On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:



Everything looks the same as another one that works except for two things.
The one that works doesn't have the "Probation passed" lines. I am not sure
if that is even part of this call.  The other is the line with "Playing
'vm-login.gsm'" in it.  at that point the working one has this:




Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.
It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if
it also uses the context 'VoiceMail', or maybe something else instead, like
'default'?).

Pete


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.
org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PBX selection

[Victor Villarreal]

Hi Speed Boy.

I agree with Emiliano Vazquez too.

Additionally, you and your team must think others points before choose
Asterisk:

* Asterisk is build to work on Linux. So your team needs some skills like
setting up a basic Linux server (Debian, Centos, etc), donwload software
from Internet, compile and install software manually.

* Your team must know how to configure Linux networking. And solve NAT
issue if apply. Basic network protocols like UDP, SIP and SDP/RDP are
welcome.

* If Asterisk needs interact with external world via VOIP provider, then
you must know how to configure SIP or IAX2 trunks. If you have analog (like
FXO) or digitals lines (like ISDN or similar), then you need ti know how to
install and configure hardware on the Linux server like telephony cards
(PCI-e or PCI) or configure VOIP gateways.

* Security: How to install and configure a basic firewall (using iptables),
o Fail2Ban. And best practices in Asterisk about this topics.

Cheers

El 17 abr. 2017 13:03, "Emiliano Vazquez" 
escribió:

> I prefer Asterisk for my projects.
>
> On Mon, Apr 17, 2017 at 11:57 AM, Speed Boy 
> wrote:
>
>>  Hi all, I'm new to VoIP, now we have a project that needs a
>>  PBX with client APPs.
>> In our team we have argument for choosing PBX. By so far, we
>>  have following candidates:
>>
>> A: Open source
>>
>>  1) Asterisk PBX (http://www.asterisk.org) (with longest
>>  history that almost every one knows it, now the last version using the
>> PJSIP stack)
>>  2) FreeSwitch (http://www.freeswitch.org) (A lot people
>>  recommended it to us)
>>
>>
>> B: Commercial
>>
>> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
>> acquired by a HongKong company now
>> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
>> also includes VoIP SDK, WebRTC and offer rebranding app for free.
>>
>> My boss prefers the Open Source PBX since they are free, but
>>  our CTO prefers the commercial editions, according to whom
>> the business PBX has better support, and the performance is
>> good, and easy to use - considering our team all are new to VoIP/PBX.
>>
>
> Hire a team with knowledge about VOIP, without your prefer if you use
> Asterisk or whatever you want
> You will win a brand new full responsibility with VOIP. The learning
> process is long and hard. You will find a lot of problems like NAT,
> intrusions. Consider learn before you pain this.
>
>
>
>>
>> We have did some searching of Asterisk, here are my questions:
>>
>> 1. Does the last Asterisk using PJSIP stack ?
>>
>
> Yes.
>
>
>> 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by
>> FreeSwicth) ?
>>
> did you google about this?
>
>
>
>
>> 3. Is it easy to compile and setup Asterisk?
>>
> You need some skills but today is really simple.
>
>
>
>> 4. Which Asterisk version is recommended? And does Asterisk support
>> Windows ?
>>
>> The latest stable release.
>
>
>
>
>> Thanks in advance .
>>
>> Best regards.
>
>
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

[Victor Villarreal]

Hi Nathan,

Personally, I create a git repo on /etc/asterisk/ folder.

With this approach, you not only can backup current dilplan on another
location (another private server, or private repo on Bitbucket account).
You can follow all the change history you made.

Simply install git, then go to /etc/asterisk/ an issue the following
commands:

#> git init
#> git add.
#> git commit -a 'First commit'

Cheers...

El 7 abr. 2017 10:48, "Steve Edwards"  escribió:

> On Thu, 6 Apr 2017, Steve Edwards wrote:
>
> You're welcome to the script at:
>>
>> http://www.sedwards.com/recover-show-dialplan.php
>>
>
> Sorry about that...
>
> Try:
>
> http://www.sedwards.com/recover-show-dialplan.txt
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager events showing in CLI

[Victor Villarreal]

Ok,

Please, check your manager.conf and logger.conf for any clue about
debugging options, into the Asterisk configuration directory.

El 26 mar. 2017 14:52, "Telium Technical Support" 
escribió:

> I tried that but it had no effect.  Still see things like:
>
>
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining
> AMI event:
>
> Event: SuccessfulAuth
>
> Privilege: security,all
>
> EventTV: 2017-03-26T13:49:39.407-0400
>
> Severity: Informational
>
> Service: SIP
>
> EventVersion: 1
>
> AccountID: 221essionID: 0x7fa0cc005cc8
>
> LocalAddress: IPV4/UDP/192.168.67.4/5060
>
> RemoteAddress: IPV4/UDP/192.168.67.26/5060
>
> UsingPassword: 1
>
>
>
>
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking
> for  Call ID: 280f68000ff289291b366a1242530ce8@192.168.67.4:5060
> (Checking To) --From tag as494dfc4b --To-tag 4155795028
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping
> retransmission on '280f68000ff289291b366a1242530ce8@192.168.67.4:5060' of
> Request 102: Match Found
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog 280f68000ff289291b366a1242530ce8@192.168.67.4:5060
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4275 __sip_autodestruct:
> Auto destroying SIP dialog 'cbf5d92f6844702b'
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog cbf5d92f6844702b
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:6138 process_message: Running
> action 'Command'
>
> [2017-03-26 13:49:39] DEBUG[1951]: manager.c:6138 process_message: Running
> action 'Command'
>
>
>
> cli> manager set debug off
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager events showing in CLI

[Victor Villarreal]

Hi Ron,

I don't remember right now, but you can try this command:

cli> manager set debug off

Cheers

El 26 mar. 2017 3:58, "Telium Technical Support" 
escribió:

I somehow cause AMI events to appear as output in the CLI, and I can’t
figure out how to turn them off.  Can someone offer a command which will
suppress AMI events/commands from showing in the CLI?



Ron



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.
org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Which tool to automatically restart Asterisk ?

[Victor Villarreal]

Hi, Oliver.

Maybe something like this (add this script to your crontab):

8<--

#!/bin/bash
#
# File: asterisk-watchdog.sh
# Date: 2015.05.26
# Build:v1.0
# Brief:Secuencia para monitorizar procesos.
#
# ${PATH}: Variable de entorno con las rutas a los ejecutables.
PATH=/bin:/sbin:/usr/bin:/usr/sbin

# ${DAEMON}: Demonio a monitorizar.
DAEMON="asterisk"

# ${MSG}: Cuerpo del mensaje a enviar por mail.
MSG="$(date '+%F %T'): ${DAEMON} se ha caido!"

pidof ${DAEMON} > /dev/null 2>&1

[ $? -ne 0 ] && { echo ${MSG}; service ${DAEMON} start; }

exit 0

--->8---

2017-02-20 11:29 GMT-03:00 Tech Support :

> Hello;
>
> Over time, we’ve built a huge enterprise level monitoring system for
> our internal and customer PBX’s. Using Nagios as the core, along with
> Grafana, Graphite, Carbon, Whisper, etc. so we can also create custom
> dynamic dashboards, we typically monitor over 1,000 different metrics for
> each PBX. For something like monitoring a system process like Asterisk,
> besides just checking to see if the process is running or not, we also
> check about a dozen or so related metrics like memory and cpu usage. If
> anything gets out of whack, the system runs the event handler to restart
> Asterisk. All the plugins are written in Perl, so they’re very easy to
> modify. What I can do if there is an interest is take the Asterisk plugin,
> strip out everything that wouldn’t apply to someone not using our system,
> and make it available to the general public. It's up to you guys. What do
> you think? Would people find that useful?
>
> Regards;
>
> John V.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Olivier
> *Sent:* Friday, February 17, 2017 10:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Which tool to automatically restart Asterisk ?
>
>
>
> Hello,
>
> Years ago, I used Monit to monitor Asterisk and restart it whenever it
> failed.
>
> Now, I wonder which tool I should pick for an Debian 8 (current) or CentOS
> 7 (future) environment.
>
> The main reason I'm looking for this tool is to avoid as much as possible,
> current 5 minutes delay between Asterisk's stop and first cutomers
> complains.
>
>
>
> 1. I always install Asterisk from source but I've read in Debian Stretch
> /etc/defaul/asterisk file, the following:
> # RUNASTSAFE: run safe_asterisk rather than asterisk (will auto-restart
> upon
> # crash). This is generally less tested and has some known
> issues
> # with properly starting and stopping Asterisk.
>
> Where I can read about those known issues ?
>
> (not found in [1]).
>
> 2. For systemd envs where /etc/init.d files are still used, what do you
> recommend ?
>
> 3. For systemd envs where /etc/init.d files are not used anymore, what do
> you recommend ?
>
> 4. Suggestions ?
>
> Regards
>
>
>
> [1] https://bugs.debian.org/cgi-bin/pkgreport.cgi?pkg=
> asterisk;dist=unstable
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Victor Villarreal]

Hi Derek,

SIP debug can be enabled via Asterisk CLI (console) with the command:

asterisk> sip set debug on

If you know via what trunk your call goes, you can use the following
command instead:

asterisk> sip set debug ip xxx.xxx.xxx.xxx

Where the xxx is the IP of your trunk (voip to pstn provider).

Affter you make all your test, simply issue:

asterisk> sip set debug off

And all the SIP conversation are saved in your full log file.

More info here:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

If what you want is test your dialplan, simply use the command:

asterisk> dialplan show xxx@your_context

Where xxx is the number you want to dial, from the context asigned to your
extension.

Cheers


El 17/2/2017 19:44, "Derek Andrew"  escribió:

> I have some troublesome numbers that I would like to capture the SIP
> dialogue when I am calling them. When I am about to dial the number, is
> there any way to turn on SIP debugging in the dial plan before I make the
> call? (and turn it off after the call is completed?)
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Disallow CALLS without registry

[Victor Villarreal]

Hi Antony,

Sory but I don't understand why your Asterisk accept anon calls with the
conf you provide us.

Maybe a full excerpt of an incoming call will help.

Last, there exist dialplan like GROUP and GROUP_COUNT that permits you
count the number of calls in a custom group fashion.

El 10/2/2017 11:51, "Антон Сацкий"  escribió:

> Thanks Frank -- but this not   a solution
> below my  current  config
>
> [general]
>
> ;sms
> accept_outofcall_message= yes
> outofcall_message_context   = messages
> auth_message_requests   = no
>
> ;general
> allowguest  = no
> jbenable= no
> jbimpl  = adaptive
> allow   = !all,g722,ulaw,gsm
> udpbindaddr = 0.0.0.0
> transport   = udp
>
> language= ru
> context = public
> alwaysauthreject= yes
> nat = force_rport,comedia
> directmedia = no
> allowoverlap= no
> match_auth_username = yes
>
> progressinband  = yes
> textsupport = yes
> videosupport= yes
> maxcallbitrate  = 1384
> ;
> sendrpid = pai
> rpid_update = yes
> pedantic=no
>  ;tos
> tos_sip=cs3
> tos_audio=ef
> tos_video=cs4
>
> 2017-02-10 16:40 GMT+02:00 Frank Vanoni :
>
>> On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote:
>>
>>
>> > so the main question is -- how to Disallow CALLS without registering
>> > on PBX
>>
>> sip.conf configuration
>> In the [general] section, define:
>>
>>
>> [general]
>> ...
>> allowguest=no
>> alwaysauthreject=yes
>> ...
>>
>>
>> The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP
>> providers connect as a guest user, however, so this may be inappropriate
>> for your situation. Also, if you want to accept anonymous SIP calls,
>> this line would block them, so you wouldn't want that. But it is listed
>> here because it is the safest configuration.
>>
>> The "alwaysauthreject" line is important. This causes a hacker to get
>> the same response from your PBX when they try to guess passwords whether
>> or not they guessed a valid username. This also has the side-effect of
>> making poorly written scanning scripts (the vast majority of hacker
>> scripts seem to be poorly written) take less resources on your Asterisk
>> box, as even if they scan a valid username, they'll think it doesn't
>> exist.
>>
>> (Source: https://www.voip-info.org/wiki/view/Asterisk+security )
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533
> tel2. +380636564340
> Paypal http://paypal.me/Satskiy
> 
> satski...@gmail.com 
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Using g729 now that patents have expired

[Victor Villarreal]

Hi Steve,

I understand your question and your point, but I use the g729 codec from
the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13
without a single problem.

So, sory but I don't share your phrase "from a lesser know web site".

About your question, I did not known that the patent has expired, so I
expect and answer just like you.

Cheers.

El 7/2/2017 19:18, "Steve Edwards"  escribió:

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards 
>> wrote:
>>
>
>   Now that the g729 patents have expired, how do we use g729 in
>>  Asterisk?
>>
>>   Will Digium be releasing a g729 codec for 'free' use or do we
>>   download the 'free' codec off the Internet now that we can use it
>>   without moral or legal restrictions?
>>
>
> On Tue, 7 Feb 2017, Carlos Rojas wrote:
>
> You can uses:
>>
>> http://asterisk.hosting.lv/
>>
>
> I'm hoping Digium will do something so we can have an 'out of the box'
> experience rather than downloading code from a lesser known web site.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk - Vtiger integration

[Victor Villarreal]

Hi Alejandro,

The documentation about your question is here:
https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager

After a few seconds of read, I think that VTigerAsteriskConnector can run
on a separate server than Asterisk PBX.

VTigerAsteriskConnector connects to Asterisk via Asterisk Manager Interface
(AMI), so you need to edit your /etc/asterisk/manager_custom.conf (because
you use Elastix distro) and create a user for the VTigerConnector. Then go
to CRM Settings -- > Integration --> PBXManager and complete all the info.

Note that seems that VTigerCoonector needs Java 1.7 onwards.

Please, follows the steps on the links. Cheers.

2017-01-13 16:04 GMT-03:00 Alejandro Cabrera Obed :

> Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
> integrate a Vtiger 6.5 server.
>
> In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.
>
> What are the requirements in the Asterisk server in order to install the
> VtigerAsteriskConnector package and then integrate the services.
>
> Thanks a lot.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Victor Villarreal]

Hi Yves,

Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of
the phone. Maybe with the snom this not happen because your switch don't
see the MAC of the Snom as a "supperted IP Phone".

2016-12-21 13:59 GMT-03:00 Yves :

> sorry... typo
> the problematic phone has the 192.168.0.13
> the asterisk has 192.168.1.211
>
> when i connect a snom phone on the cable that was in the soundstation 6000
> before and configure the
> phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...
>
> it would be helpful if someone, that has a running soundstation ip 6000
> could send the configuration... :-/
>
> regards,
> yves
>
>
>
> Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:
>
>> On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:
>>
>>> Hi Mark,
>>>
>>> yes, you are right... these are different VLANs
>>> I configured the other phone to use the same IP (192.168.1.13)... and it
>>> worked flawlessly... on the SAME Networkcable in the same plug...
>>> so it must have something to do with the polycom phone config...
>>> remember...
>>> when I use tcp the phone tries to register, but does not even try with
>>> udp...
>>>
>>> thank you,
>>> yves
>>>
>>>I am a bit confused: is your problematic phone's IP 192.168.0.13
>> (what the error log is reporting below) or 192.168.1.13?
>>
>> Am 21.12.2016 um 13:34 schrieb Mark Wiater:
>>>
>>> Yves,
>>>
>>> Didn't you say that
>>>
>>> AsteriskServer: 192.168.1.211
>>> SIP-user: 165
>>>
>>> ?
>>>
>>> On 12/21/2016 4:24 AM, Yves wrote:
>>>
>>> . It is sure for 100% that there is no firewall or something else
>>> mangeling
>>> in between... another Hardphone works as expected using the same
>>> Netzworkcable on the same Networkplug with UDP on Port 5060...
>>>
>>>
>>> This other hardphone, what IP does it have?
>>>
>>>
>>> 50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
>>> 255.255.255.0
>>>
>>> The line above suggests to me that your phone and your asterisk server
>>> are
>>> on a different network, there has to be something that routes between
>>> those
>>> two networks. Often what routes, can firewall.
>>>
>>> 000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
>>> Temporarily not available
>>>
>>>
>>>
>>> Mark
>>>
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Victor Villarreal]

With all the money you plan to invest in firmware, licenses, etc., you have
bought a Grandstream IP phone or Yealink...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] iowait issues on CentOS 7

[Victor Villarreal]

Hi Luca,

IO delay maybe come from Hard Disk lattency. You can exec an "lsof "
command to view what file asterisk proccess hold down when load spike.

If there are some call recording, you can configure Asterisk to make it in
a temp location, a RAM Disk in Linux.

If you make hard usage of the AstDB file, you can copy it to RAM too, to
avoid read/write to the disk.

Please, read this post about lsof:
http://0xfe.blogspot.com.ar/2006/03/troubleshooting-unix-systems-with-lsof.html

You can view something weird in Asterisk logs when high load ? Maybe enable
debug ?

You can install and setup "atop". Then you can review the system status
after the load peak and drill down, just to the problem.

Finally, for troubleshooting IO Wait on a Linux system, you can view this
post: http://bencane.com/2012/08/06/troubleshooting-high-io-wait-in-linux/

Cheers

2016-11-23 12:32 GMT-03:00 Luca Pradovera :

> Hello!
> One of our customers has  an issue where our load average on two of the
> boxes spikes on peak loads. What I got from testing is consistent with what
> they were reporting: on CentOS 7, the load spikes in hockey stick fashion,
> from 40-50% up to 200%, with very high iowait values.
> On CentOS 6, load increases and decreases linearly and the machine never
> slows down.
>
> Asterisk version is 1.8.22.0, which is if course quite old (but it is what
> is installed).
> The CentOS 6 box actually has less RAM (8 Gb vs. 16 Gb), but other than
> that they are exactly identical in hardware configuration.
>
> I checked the usual culprits, but to no avail. Is this a known issue?
>
> Best regards,
>
> Luca
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] iaxmodem errors.

[Victor Villarreal]

Hi John!

I'm not sure why are you using iaxmodem... I use it  a few years ago with
Asterisk 1.4

In Asterisk v11 fax is managed  using res_fax. Please see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReceiveFAX_res_fax

You only need download, compile and install the spandsp lib for your distro
(fax depend  on it) and then recompile Asterisk (if you don't have this
resource module already).

We currently receive fax via g.711 ir SIP (via T.38), convert it into PDF
and send via email. All  with Asterisk v11 and OpenSource software. I can
send you our scripts if you want.

Cheers
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Force hangup not working on stuck channel

[Victor Villarreal]

Hi Carlos,

Did you try with the following CLI command:

CLI> channel request hangup  CHANNEL_NAME

???

El nov. 3, 2016 1:16 PM, "Carlos Chavez"  escribió:

> I am unable to force a hangup on a channel that has been stuck for over
> two days:
>
> IAX2/from-CD-11006   oficina  27701 Up
> Dial IAX2/to-CD/2883   3467130007  46:24:59 Sotelo
> Sotelo  IAX2/to-CD-20713
>
> I have tried "hangup request IAX2/from-CD-11006" several times but no
> joy.  I also see the following in the CLI:
>
> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
> much delay in IAX2 calltoken timestamp from address X.X.X.X
>
> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old
> but new client so haven't had time yet to upgrade to 13).  Because this
> channels is stuck
>  all other calls between servers are not working.  The only way I have
> found to resolve the problem is to stop and restart Asterisk.  This is
> obviously a great inconvinience so is there a way for force iax to unload
> even if there are channels in use?  Or any other way to kill these stubborn
> channels?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)9116-91161
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Victor Villarreal]

Ok.

Please, note that 192.168.1.37 (I suspect) is the internal  LAN address Of
the Polycom hardphone. If this is true, then you have  NAT issues.

The REGISTER message are received by your PBX, but when respond, Asterisk
send the next SIP message to the IP informed by the phone, that is the
internal LAN address. The messages do not reach back to the hardphone.

You need to setup a STUN server in the Polycom hardphone settings. Please,
check the manual. Search in Google some public  STUN server to put in the
settings.

Last, the idea behind the "sip set debug" command was view the complete SIP
messages conversation, not search for an error.

On NAT escenarios, remember:

* The NATed phones need to know the public  IP of the NATing router. Either
by manual setting  or  by STUN protocol.

* Reduce the time between REGISTERs attempt, if the client  have a dynamic
IP connection.

* Use the "localnet" SIP settings in Asterisk, so the PBX can distingish
what Network need contacted via NAT and what not.

Cheers.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Victor Villarreal]

Hi Motty,

Please, set  Verbose  to 3 and Debug to 3 At Asterisk CLI. Then "sip set
debug on".

Now try to register again. At last, " sip  de debug off".

Examine tour console  or  full log file to find some clue ir send me back
some trace.

Cheers.

El oct. 13, 2016 1:45 PM, "Motty Cruz"  escribió:

> Hello, fresh install of Asterisk 13.11.2, client unable to register.  For
> now I have IPtables disabled, also selinux is disabled
>
>
>
> [1006]
>
> type=friend
>
> username=1006
>
> secret=mysecret
>
> context=sip-phone
>
> call-limit=1
>
> callerid="iuser" <1006>
>
> disallow=all
>
> host=dynamic
>
> allow=all
>
>
>
> any ideas?
>
>
>
> Thanks,
>
> Motty
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

[Victor Villarreal]

Hi Jonas!

Do you currently use any TLS technology in your Asterisk? Like SIP-TLS o
pjSIP-TLS support ? If don't, please go to modules.conf and start disabling
some modules that you don't use.

For example, I can see some other modules related to calendars. If you
don't use this, please disable it. You gain a lower memory footprint, and
maybe fix your issue.

I hope this help you. Cheers.

2016-10-11 9:41 GMT-03:00 Jonas Kellens :

> Hello
>
> I am experiencing a freeze of the Asterisk proces when issuing a 'sip
> reload'.
>
> I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
> 13.10.0 and certified-13.8-cert3.
>
> I do not have this on versions certified-13.8-cert2, certified-13.8-cert1
> and asterisk 1.8.32.3.
>
> The only solution is a cold restart of Asterisk.
>
> I can execute any command on CLI except 'sip reload'.
>
> This is what I have on CLI :
>
> sip5*CLI> sip reload
> [Oct  7 23:58:40]  Reloading SIP
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/sip.conf': Found
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/sipTemplates.conf': Found
> [Oct  7 23:58:40]   == Parsing '/etc/asterisk/users.conf': Found
> [Oct  7 23:58:40]   == Using SIP TOS bits 96
> [Oct  7 23:58:40]   == Using SIP CoS mark 3
> [Oct  7 23:58:40]   == TLS/SSL ECDH initialized (secp256r1), faster PFS
> cipher-suites enabled
> [Oct  7 23:58:40]   == TLS/SSL certificate ok
>
> --> no more output on CLI. Asterisk has gone completely !
>
> Another 'sip reload' gives :
>
> sip5*CLI> sip reload
> [Oct  8 00:01:10] Previous SIP reload not yet done
>
> sip5*CLI> sip reload
> sip5*CLI>
>
>
> Other commands are no problem on the CLI (while the freeze occurs ! ) :
>
> sip5*CLI> core show  version
> Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a
> x86_64 running Linux on 2016-10-07 21:27:15 UTC
>
>
> sip5*CLI> sip show channelstats
> Peer Call ID  Duration Recv: Pack  Lost   ( %)
> Jitter Send: Pack  Lost   ( %) Jitter
> 0 active SIP channels
>
>
> sip5*CLI> core show threads
> 0x7f97ff0fb700 2849 netconsole   started at [ 1639] asterisk.c
> listener()
> 0x7f97fe843700 2760 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff367700 2759 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97fe8bf700 2758 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c
> asterisk_daemon()
> 0x7f97fe9b7700 2172 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fea33700 2171 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97feaaf700 2170 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97feb2b700 2169 scan_thread  started at [  920] pbx_spool.c
> load_module()
> 0x7f97feba7700 2167 cleanup  started at [  400] pbx_realtime.c
> load_module()
> 0x7f97fec23700 2165 lock_broker  started at [  524] func_lock.c
> load_module()
> 0x7f97fee13700 2161 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fec9f700 2164 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fed1b700 2163 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fed97700 2162 cal->tech->load_calendar started at [  489]
> res_calendar.c build_calendar()
> 0x7f97fee8f700 2160 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fef0b700 2159 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97fef87700 2158 default_tps_processing_function started at [  200]
> taskprocessor.c default_listener_start()
> 0x7f97ff003700 2157 do_monitor   started at [11645] chan_dahdi.c
> restart_monitor()
> 0x7f97ff07f700 2156 do_monitor   started at [29518] chan_sip.c
> restart_monitor()
> 0x7f97ff1f3700 2153 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff2eb700 2151 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff26f700 2152 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff3e3700 2149 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff45f700 2148 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff4db700 2147 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff5d3700 2145 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff557700 2146 worker_start started at [ 1077] threadpool.c
> worker_thread_start()
> 0x7f97ff64f700 2144 worker_start started at [ 1077] threadpool.c
> 

Re: [asterisk-users] send a call to moh until user is available

[Victor Villarreal]

Hi Tux John,

The behavior you need is cover in Asterisk within a Queue.

1. Create a new queue in queues.conf and assign as static member, the 4450
extension.

2. In your dialplan, you need to route the incomming calls to the new queue
and pass as argument the timeout in seconds you want when hangup the
waiting calls.

When a new call arrives, it enter the queue. The Callee ear moh music,
while the 4450 ring if its available. Ir not, the queue system wait until
the 4450 become available, an then send the call.

Please, refer to http://www.asteriskguru.com/tutorials/queues.html

The #3 title (simple queue) is all you need.

Cheers
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 147, Issue 5

[Victor Villarreal]

Hi all ! Thanks for your feedback and sory for the delay. Respond:


> Date: Mon, 3 Oct 2016 21:05:55 -0300
> From: Marcelo Terres 
>
> I think that you need the dev files too. In Debian 8, the package is
> libmysqlclient-dev.
>
> But Debian 8 uses libmysqlclient-18. Where did you get the 20 ?
>
> Regards,
>
> Marcelo H. Terres 

Hi Marcelo,

My idea was to install a new PBX to one of my clients, but with the latest
mySQL version.

I follow these instructions: http://dev.mysql.com/downloads/repo/apt/

The libmysqlclient-dev package is installed, but from the mysql repo. It's
version 20.



@Tzafir, thanks for your reply. I respond you inline...

> Date: Wed, 5 Oct 2016 17:13:41 +0300
> From: Tzafrir Cohen 
>
> For the record, we ubild both asterisk 11 (last version: 11.21.2) and 13
> (13.11.2) for Debian Stable using the distro-provided MySQL packages.

Ok, I take note of this. My idea is install Asterisk-11 from source instead
of use a "distribution package".
It's not the first time I compile Asterisk (in fact I work with Asterisk
from v1.4 in production
on many mission critical projects with greats results). But this time I
decided to use the latest mysql version
and face a problem that exceed my knowledge scope :-(

> Is Are there any mysql-related module loaded?
>
> Start with e.g.
>
>  ldd /usr/lib/asterisk/module/cdr_mysql.so

No, there is no mysql-related module compiled or loaded. Only ODBC:

sistemas@nodo1:~$ ls -lh /usr/lib64/asterisk/modules/ | grep mysql

sistemas@nodo1:~$ ls -lh /usr/lib64/asterisk/modules/ | grep odbc
-rwxr-xr-x 1 root root 331K oct  3 17:36 cdr_adaptive_odbc.so
-rwxr-xr-x 1 root root 265K oct  3 17:36 cdr_odbc.so
-rwxr-xr-x 1 root root 339K oct  3 17:36 cel_odbc.so
-rwxr-xr-x 1 root root 382K oct  3 17:36 func_odbc.so
-rwxr-xr-x 1 root root 325K oct  3 17:36 res_config_odbc.so
-rwxr-xr-x 1 root root 350K oct  3 17:36 res_odbc.so

It's like the compile script don't found my v20 libmysqlclient package
installed.
What is the routine responsible for this job?

Asterisk here in this server, was compiled with a libmysqlclient v18, but I
unistalled this package now
because Asterisk-11 compiled with v18 and connected to a v5.7 mySQL
instance, return an Asterisk crash on libmysqlclient.so module.

If it's don't possible to compile Asterisk-11 against libmysqlclient-20, I
will have to downgrade my mySQL-5.7 instance back to 5.5 Debian version :(

Any idea?

Thanks in advance and best regards.

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

[Victor Villarreal]

Hi List!

I'm facing a problem while compiling Asterisk-11 on a Debian 8 server.

The mysql-server version installed is 5.7 and come from the official mySQL
community repo for Debian.

After compile, install and execute Asterisk, the comman "lsof -p `pidof
asterisk` | grep mysql" don't produce any output. Like if confgure script
don't found the mysql lib.

With libmysqlclient18 every is Ok. How can I use libmysqlclient20 with
Asterisk ?

Thanks in advance, and best regards.

root@nodo1:/usr/src/asterisk-11.23.0# ls -lh /usr/lib/x86_64-linux-gnu/ |
grep mysql
-rw-r--r-- 1 root root 5,7M ago 25 09:37 libmysqlclient.a
lrwxrwxrwx 1 root root   20 ago 25 09:37 libmysqlclient.so ->
libmysqlclient.so.20
lrwxrwxrwx 1 root root   24 ago 25 09:37 libmysqlclient.so.20 ->
libmysqlclient.so.20.3.2
-rw-r--r-- 1 root root 4,2M ago 25 09:37 libmysqlclient.so.20.3.2
-rw-r--r-- 1 root root  18K ago 25 09:37 libmysqlservices.a

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to read sip debug

[Victor Villarreal]

Hi Thufir,

The analysis of a SIP Debug depends on what the problem to be solved.

If you experience problems with inbound calls from a SIP trunk or
provider, you can type in Asterisk cli 'core set debug 3' and then
'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP
provider or from where it is supposed to come call.

Then you make a test call, and look in full log an INVITE message
(note that you analize an OPTION message in your mail, but I think
that this not help in this case).

After the incoming INVITE message from your SIP provider, you can
follow the rest of the Asterisk logic and look for the reason why
Asterisk is denying that call.

Hope this help you.

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

[Victor Villarreal]

Hi List,

I solve this issue and I want share it with this community.

The sng-tc-linux-1.3.8 package don't compile across Certified Asterisk.
Only normal Asterisk like 11.22.0 version.

We have this version in production with the D100 board. Working.

Cheers

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Identify more demanding routine inside Asterisk

[Victor Villarreal]

Hi List !

I'm facing a problem with the CPU consumption in Asterisk 11.22.0.

I could decrease a lot of load, migrating both the astdb.sqlite3 and call
recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and
nodiratime flags), manually spread each of the hardware interrupts (network
interfaces, wanpipe and megasas) to an individual dedicated CPU core and
stick the Asterisk process to other dedicated CPU core (free of hardware
interrupts).

Now the usage is 90% of his core with 300 active channels (66% SIP + 17%
IAX + 17% DAHDI). 17% of channels are g.729 transcoded via software. Some
considerations are that there is +30 AMI concurrent clients, plus SIP
Realtime peers. I don't have any AGI script or ODBC custom function/query.
The DialPlan is minimal for the work it does too.

My question is: Is there any way to identify the more demanding
routine/task in Asterisk so I can know where to tweak ?

I read the excelente post of Moy [1], but the most usage of CPU is in the
mail process of Asterisk, not in any of the threads. Moreover, the pstack
command don't work on my 64-bits system.

Hope anyone can help. Cheers.

[1]
https://moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] nagios asterisk check SIP

[Victor Villarreal]

On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote:
> Iam loocking for an programm to check the SIP port of an Asterisk
asterisk.
>
> Ome time ago I have used
> #/usr/bin/sipsak
> but it seemed that it is not working anymore?

Hi Thomas,

Maybe this links help you:
http://fabian-affolter.ch/blog/nmap-scripts-for-voip-analyses/

Not for sipsak, but for great nmap.

Cheers

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue_log - odbc vs AMI

[Victor Villarreal]

Hi Marek,

Here, we have an Asterisk v11-cert11 and found that there is NOT equal the
CDR via AMI and CDR in Database.

Please, check my gist:
https://gist.github.com/MefhigosetH/89462e599a996dedf048f8d2b4e94d47

We have in use some custom dialplan variables in CDR (ie.: groupcount and
rptqos), and these variables are visibles in CDR table BUT ARE NOT SHOW in
AMI Event.

Hope this help. Cheers from Argentina.


-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SPA112 flapping

[Victor Villarreal]

Hi Mike,

I would try the following:

* If you can login through HTTP, check the uptime of the Cisco device. Make
sure the device is not rebooting.
* If you can, make a 'ping' from the PBX to the device and annotate
milli-seconds of response. Then compare then to the default 'qualify' sip
setting for the Cisco peer (width sip show peer _SPA112_PEER_NAME_). Maybe
you can set 'qualify=X' where 'X' is the measured round-trip time to a peer.
* If the Cisco is behind a NAT device/router, maybe the default 60 seconds
for the 'qualifyfreq=60' sip setting is not enough to keep active the
session. Try changing this value to something lower like 15 or 30 seconds.

Cheers

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

[Victor Villarreal]

Hi there !

Someone in this wonderful list tried to install Sangoma transcoding board
D100  on Asterisk v11 ?

I followed each of the steps in the wiki [1], but when running 'make
asterisk' receipt compilation errors about the absence of some header files
[2].

I exchanged some mail with the official support, but still I have not
received any solution.

Using the following software version:

* Debian 7.11 with up-to-date packages.
* Linux nodo3 3.2.0-4-amd64 #1 SMP Debian 3.2.78-1 x86_64 GNU/Linux
* certified-asterisk-11.6-cert11 compiled from official Asterisk-org source
code.
* sng-tc-linux-1.3.8.x86_64

TIP: I've tried compile again through v11.22.0 source code with similar
resoults.

Thanks in advance, and best regards.

[1] http://wiki.sangoma.com/Asterisk-D100-Single-Server-Installation
[2] https://gist.github.com/MefhigosetH/883589726c52b8dc72f3cfd6825fe3f1

-- 
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

[Christian Victor]

Hi James,

we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
machine with quite heavy line usage. No codec conversion course.

I don't believe that there is a hard limit of E1s coded into Asterisk.
But the maximum lines you can squeeze out of your specific hardware
depends on so many factors (asterisk version, hardware etc).

Chris

2010/3/25 James Lamanna jlama...@gmail.com:
 Hi,
 Does anyone have any good empirical data suggesting what the maximum
 number of PRI calls (incoming and outgoing)
 without hardware echo cancellation can be handled on a single box is?
 I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
 D-Channels going down and then coming back up (See below).

 I've looked at the number of simultaneous calls at each of these
 points, and each time the span seems to
 have around 21-23 calls, and the total number of calls ranges between 47 and 
 53.
 I'm trying to figure out if this is a load issue or an issue on the
 provider side, though my provider says they
 do not see any errors on any of the T1s.
 Could this be some sort of hardware interrupt problem? If so, how can I check?

 The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT)
 4GB memory.
 Running asterisk 1.4.26.3 (32-bit)
 with libpri-1.4.7 and zaptel-1.4.12.9

 Thanks.

 -- James

 Please CC me on responses.


 [Mar 22 09:45:00] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 
 down
 [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available!
  Using Primary channel 48 as D-channel anyway!
 [Mar 22 09:45:00] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 up
 [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
 down
 [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
 [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
 [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
 down
 [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
 [Mar 22 09:59:23] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
 [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
 down
 [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
 [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
 [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
 down
 [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
 [Mar 22 10:36:11] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
 [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad
 FCS (8) on Primary D-channel of span 3
 [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad
 FCS (8) on Primary D-channel of span 1
 [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort
 (6) on Primary D-channel of span 2
 [Mar 22 11:30:53] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 
 down
 [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
 [Mar 22 11:30:53] VERBOSE[8886] logger.c:   == Primary D-Channel on span 1 up
 [Mar 22 15:34:28] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 
 down
 [Mar 22 15:34:28] WARNING[8887] chan_dahdi.c: No D-channels available!
  Using Primary channel 48 as D-channel anyway!
 [Mar 22 15:34:28] VERBOSE[8887] logger.c:   == Primary D-Channel on span 2 up

 --
 _
 -- 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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

[Christian Victor]

2010/3/25 Steve Edwards asterisk@sedwards.com:
 On Thu, 25 Mar 2010, Tzafrir Cohen wrote:

 [snipping a lot of interesting technical and historical details]

 As you can see, there's actually a limit at the DAHDI level.
 DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is
 1024. That's as many channels that you can have.

 All good stuff. I would just question the sanity of putting 1024 eggs in a
 single basket.

Well - think of having 2048 eggs and only two hands. ;-)

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

[Christian Victor]

2010/3/25 Zeeshan Zakaria zisha...@gmail.com:
 Tzafrir, so you have actually worked with more than 192 concurrent zap
 channels, which means more than 8 spans, on a single server, and can verify
 that it actually works without freezing asterisk.

 As I have written before - I did use 8 E1 in one machine quite often.
So I can confirm that it runs with at least 240 B-channels. If my mind
serves me right we even had one installation with 480 channels - not
with TE410P though.

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] G.711a or G.711u ???

[Christian Victor]

2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com:
 Dear all, I have an Asterisk SIP server in a LAN environment and I want your
 opinion in order to decide the use of an audio codec:

 What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
 calls ???

That depends most on the codec used in your country IF you terminate
calls to ISDN. In th USA (and some other countries) you would use
G711u and in Europe etc. you would use G711a.

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] register = 2345:passw...@sip_proxy/1234

[Christian Victor]

2010/3/19 tjoen tj...@dds.nl:
 register = tjoen:mypas...@sip_proxy/1234

 [sip_proxy]
 type=peer
 host=ekiga.net

I guess you need to register to the actual hostname, not the peers name.

register = tjoen:mypas...@ekiga.net/1234

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digium TE4xx T1 Bonding

[Christian Victor]

2010/3/11 Eric Wheeler aster...@ew.ewheeler.org:

 4. Does anyone have a couple TE2xx or TE4xx cards that can test such a
 configuration? I would like to research their capability before
 purchasing a couple $1200 cards.

Hi Eric,

I have four spare TE411P but never used bonded T1 or T1 for data at
all. If you can supply me with instructions I could test if your plan
works the way you need.

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hardware requirements question.

[Christian Victor]

Yes, this machine will be enough for that task. Performance wise. The
other good thing is that it is not very likely that someone will steal
your PBX. As far as I remember it is a 7 rack unit box which weights
approx. one metric ton. ;-)

But remember - if anything dies in the box and you have to get spare
parts quick you will pay more than you want to.

Chris

2010/3/5 David Little da...@mandm-tech.com:
 I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors,
 SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop
 an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP).
 I also will install a sound card for an intercom. Is this hardware
 sufficient if  using a Digium TDM2400P?

 --
 Thanks,

 David Little
 MM Technology, Inc.

 da...@mandm-tech.com
 704.882.9432 x3
 704.882.0405 FAX


 --
 _
 -- 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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Playback in h extension

[Christian Victor]

2010/3/5 Danny Nicholas da...@debsinc.com:
 Not possible.  H exten is called by a hangup.

Well - sometimes not both parties hang up at the same time. ;-) If you
want to play something to the originating party after die Dial()ed
party hangs up use the option g in the Dial command to get more
commands executed after the called party hangs up. There you could
check the system variable DIALSTATUS to check if the called party
ANSWERed the call or was BUSY etc.

I hope that helps a bit. I just wrote it from the back of my mind.
Please check the documentation of the Dial command.

If you are not in a Dial() situation Danny's comment applies. ;-)

Chris

-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Do i need install Dahdi or libpri ?

[Christian Victor]

2010/2/25 Zhang Shukun bit...@gmail.com:
 next ,i want to dial from asterisk to PSTN now. i have see the sample
 in the extensions.conf relevent to PSTN as follow:

 ; If you are freely delivering calls to the PSTN, list them here
 ;
 ;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
 ;exten = _1256325,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

I am pretty sure you need to dial through SIP technology not Dahdi.
Like Dial(SIP/telephonenum...@your-gateway-ip)

 but above shows something about DAHDI card.

Wich you don't have.

 my question is:

 a, Do i need install DAHDI or libpri in my system?

Not to connect your Gateway through sip.

 b, how to write in dialplan to realise connection to PSTN.

Thats a quite general question. But I guess the Dial command above
will lead you the right way.

Chris

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 4 PCIe cards in one asterisk server

[Christian Victor]

Not wit four - but two of them in a single core 3GHz machine worked
flawlessly doing only switching and IVR without codec conversion.

Many will suggest that you split your lines on two machines to to
prevent a total loss when a machine fails. This will add some work on
setup but maybe save you some worries.

Christian

2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl:
 Hi,



 Does anybody have any experience with asterisk where are four PCIe cards are
 used in one server (TE420).

 So you can have max 4 * 4 * 30 channels = 480 channels used.



 Regards,



 Arjan Kroon

 Mobillion BV



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Question regarding digital card TE412p

[Christian Victor]

Hi!

Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.

Chris

2009/12/14 das sandesh sandesh...@gmail.com:
 Hi,
 I was able to implement T122p one port PRI and was able to call out, but I
 am planning to use TE412p (includes echo cancellation) 4 port digital card
 (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
 connections) with proper hardware like dual core quadcore processor and 8gb
 RAM in one server?
 Also I was planning to implement using 64 bit architecture with Asterisk:
 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2.
 Thank you very much for your help.
 Regards
 Sandesh
 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] G729 Pass through

[Christian Victor]

Hi!

Are you sure you are getting Astrisk out of the media path? I guess
reinvite must be allowed. Then it should work without transcoding
licenses.

Maybe you should take a look at the SIP DEBUG info to see what codec
Asterisk is trying to negotiate with the trunk. You could disallow
alaw and ulaw for a test.

Christian

2009/12/11 Dovey Forman dovey.for...@idt.net:
 Hi;



 I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
 endpoints.



 It seems that when I enable G729 on my peers in sip.conf and make a call I
 am getting the following errors:



 Called crp_uk/806575011971553141421

 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation
 path from g729 to ulaw

 Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256,
 while native formats is 4 (read/write = 4/4)



 Both my end points (Aastra phone) and my sip carrier support G729, so this
 should be simple pass-through.



 Snippet of my peer crp_uk:



 [crp_uk]

 disallow=all

 allow=ulaw

 allow=alaw

 allow=g729

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma A101DE with Dell PE 2850

[Christian Victor]

2009/12/8 Ricardo Melendez rmelen...@utep.com.mx:
 First I see at sangoma page that A101DE is PCI-Express  (I think  x1 for the
 size of the connector)

Yes, it is PCIe x1. There is an A101D wich is PCI(-X).

 for PCI Express

 one x4 lane width
 one x8 lane width

 I can connect the card to any of the slots?, or only to PCI-Express Slots?
 (is compatible the card with x4 and x8 PCI-Express slots?)

Yes, the A101DE runs in PCIe x4 or x8 and the A101D will run in PCI or PCI-X

Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] automon = *1 one touch recording

[Christian Victor]

2009/12/8 Joseph syscon...@gmail.com:
 After pressing *1 console is not showing anything indicating that the call 
 is being recorded:

 -- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0, 
 transfer) in new stack
     -- SIP/479-1270-680060b0 Playing 'transfer' (language 'en')
     -- Executing [...@office-closed:2] Dial(SIP/479-1270-680060b0, 
 SIP/11IAX2/iaxy-322|30|rwW) in new stack
     -- Called 11
     -- Called iaxy-322
     -- Call accepted by 10.0.0.108 (format ulaw)
     -- Format for call is ulaw
     -- SIP/11-007855f0 is ringing
     -- IAX2/iaxy-322-6005 is ringing
     -- SIP/11-007855f0 answered SIP/479-1270-680060b0
     -- Hungup 'IAX2/iaxy-322-6005'
   == Spawn extension (office-closed, 11, 2) exited non-zero on 
 'SIP/479-1270-680060b0'

Did you make sure that your telephone actually sends the DTMF tones
(the right way)? It seems that asterisk does not recognise incoming
DTM or your verbosity level is not high enough.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Pbx-cards

[Christian Victor]

mattias schrieb:
 But are not pbx card and modem the same?

There are single FXO cards (to connect to a analogue line) that are
basically PCI modem with a special driver. But the chances that your
modem is compatible to this one specific type is very little.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.4 and Fax

[Christian Victor]

2009/11/2 Doug Lytle supp...@drdos.info

 Dan Journo wrote:
 
  I need to get it up and running before we can put in the order to
  transfer the fixed line number over to SIP.
 

 Faxing over SIP is never a good idea.


And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.4 and Fax

[Christian Victor]

2009/11/2 Doug Lytle supp...@drdos.info

 Christian Victor wrote:
  2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info
 
 
  Faxing over SIP is never a good idea.
 
  And why would that be? I think that faxing over SIP using T.38 is a
  fantastic idea.

 As far as I know, T.38 isn't supported under 1.4


That would be Faxing using Asterisk 1.4 is never a good idea. Sorry for
being such a bean counter. ;-)

To stay on-topic: Terminating fax over PSTN works quite well in 1.4 but the
original poster should be warned of trying to terminate fax over a SIP
trunk. Using SIP/G.711 to connect the fax machine to Asterisk over LAN works
quite well in my experience but others had worse results.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] GSM cellphone as cheap gateway?

[Christian Victor]

Olivier schrieb:
 2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
 
  There are FTC’s available,

 What is it (a FTC) ? a cable ?
 Any pointer to that (Google is helpless)? ?

My guess would be fixed to cell or FX to cell adapter.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] All the four lights blinking

[Christian Victor]

2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com

 Thanks you very much Kevin.I will try it by connecting one end of
  Ethernet cable to one slot and other to second slot . Configuring one
 as pri_net and the other as pri_cpe.

 I will provide you feed on monday either i succed or not

 Remember that you CANT NOT use an Ethernet cross-over cable. You need to
get a E1 cross-over cable. Google for the pinout.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best ISDN BRI solutions?

[Christian Victor]

2009/8/6 Alex Balashov abalas...@evaristesys.com


 Sure it is.  Just get a media gateway that does T.38 - and does it
 relatively well.


Wich the Pattons do quite well afaik.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk]q: asterisk 1.6.1 install

[Christian Victor]

tom schrieb:
 hi
 just donwloaded the 1.6.1 branch and made configure  install. so far 
 so good. after staerting asterisk with:

 asterisk -cr
 Could not load features.conf
   == Registered application 'ParkedCall'
   == Registered application 'Park'
   == Manager registered action ParkedCalls
   == Manager registered action Park
   == Manager registered action Bridge
   == Manager registered action DBGet
   == Manager registered action DBPut
   == Manager registered action DBDel
   == Manager registered action DBDelTree
  Asterisk Dynamic Loader Starting:
 No 'modules.conf' found, no modules will be loaded.
 Asterisk Ready.
 CLI
Did you make samples?

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Different codecs for reading and writing

[Christian Victor]

Philipp Kempgen schrieb:
 Elliot Murdock schrieb:
   
 I am wondering how the Asterisk community has been working on
 solutions to deal with the asymmetric quality of ADSL.   Voip is
 becoming popular and a bottleneck does exists on the ADSL upload side.
 

 One participant's upload is the other participant's download and
 vice-versa. So how would different codecs for sending and receiving
 help?
   
Given that the other party does not use an asymetric internet connection 
it could actually help. Not that I would recomment such a mode over just 
using a low bandwidth codec like the aforementioned.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk

[Christian Victor]

Jeff LaCoursiere schrieb:
 I have a question in to them about how that floating licensing works, 
 though.  Does that mean that with every call a license check must be made? 
 I don't see how it would work otherwise, and that means my whole business 
 - every call - is dependant on their license server being up and 
 reachable.
I guess that you run your own license server and your machines check the 
availability of one of your licenses there. At least thats how some 
other companies for e.g. TTS licenses do it.

   I also don't think that the slight added convenience is then 
 worth the recurring cost annually.  The price of the license is comparable 
 to Digium in US dollars.
   
If you are running a couple of servers and you don't know where your 
G729 calls will arrive then it makes sense to me.

If you run G729 only and have licenses for every line in your system 
then it obviously makes no sense. But if you have for example 1200 lines 
and 10% G729 users you never know if they are spread over all your 
server or all arrive on one machine.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Question about core CDR system for multilpe servers

[Christian Victor]

Danny Nicholas schrieb:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
 Sent: Thursday, June 04, 2009 11:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Question about core CDR system for multilpe
 servers

 Gustavo A Gonzalez escribió:
   
 Hi all! I’m not sure if it is the correct place but, I’ve five boxes
 
 running
   
 asterisk and each one with his own cdr mysql database. What Im looking for
 is to get a core CDR system that holds information stored on each asterisk
 server. Have you any suggestion/process to accomplish that?. Thanks!!!

 Gustavo A. González

   
 
 Well, this sounds fairly simple. Can you do it by configuring each 
 asterisk server (cdr_mysql.conf) to connect to the same MySQL core 
 database server. Inside it, you can have each server CDR in a separate 
 database, or in a single database for all of them using different table 
 names. How to configure it, depends on performance inside the MySQL 
 server, and how do you want to store the information. Maybe is not a 
 good idea to have all the CDRs on the same database if the tables are 
 going to be too big. But having all of them in a single database server, 
 shouldn't be a problem.

 Cheers,

   
 This all sounds very nice and do-able, but doesn't this sound like a
 
 high-odds scenario for creating a single point-of-failure especially 
if the

5 machines are all creating a high volume of calls?

True - so use a MySQL cluster instead.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IP phone recommendation

[Christian Victor]

Right! Whatever somebody likes more! I just say that the Snoms look 
better at the side of my Mac. Wich is of course by far the superiour 
system. ;-)

Chris

John Novack schrieb:
 Hasn't this religious argument/discussion gone on long enough??


 zoach...@securax.org wrote:
   
 I personally find the snom phones to look quite good compared to the 
 american and chinese brands, might be a european thing though :)

 Zoa


 Geraint Lee wrote:
   
 
 I personally find the snom phones to be generally ugly and 
 un-finger-friendly, in terms of reliability and quality, never had 
 any trouble, good phones all in all, i just can't get past the tacky 
 look and feel so don't buy them.

 2009/6/3 Darrick Hartman dhart...@djhsolutions.com 
 mailto:dhart...@djhsolutions.com

 On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote:
 
  On Thu, 4 Jun 2009, Rob Hillis wrote:
 
  Jeff LaCoursiere wrote:
  We are still talking about a $175 phone.  How about the
 Polycom IP 320?
  $85 at 888voipstore.  Can't go wrong with Polycom for voice
 quality.
 
  True, Polycom's are brilliant for voice quality, but unlike the
 Snom, a
  Polycom /will/ reboot on the drop of a hat /and/ take a damned
 long time
  to do it (~45-60 seconds)  In addition, the web interface should be
  taken away and shot - the only real way to configure them is
 through (T)FTP.
 
  They are however, extraordinarily configurable through the XML
 config
  and they are very stable.  Once they're configured they work very
  nicely.  The lack of a decent number of BLF keys (even with a very
  expensive sidecar you only get two more keys than a standalone
 Snom320)
  puts me off a little.
 
  However, for a conference phone, the Polycom's can't be easily
 beaten.
  Their handsfree call quality is in a league of it's own.
 
 
  Mainly I suggest it because the OP asked for an inexpensive
 quality phone.
  I agree on the provisioning - the web interface is useless, and
 unless you
  know how to setup the XML files properly you are doomed to a very
  frustrating experience.

 The Polycom 320/330's are nice little phones for the price.

 There are several resources for configuring the phones from the XML
 config files.  If the config files are sane, the phones don't take
 that
 long to reboot.

 This is probably one of the better examples:

 http://www.kfife.com/voip/

 Karl did a good job commenting in the config files where he made
 changes.

 Darrick

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


 __ Information from ESET NOD32 Antivirus, version of virus 
 signature database 4124 (20090602) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com

   
 
   

 __ Information from ESET NOD32 Antivirus, version of virus signature 
 database 4124 (20090602) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

   
 

   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FritzBox 7270

[Christian Victor]

Manoj Panicker - FOES schrieb:
 However I can always call any one pre-configured PSTN number using the
 call forwarding feature, however I should be able to use my sogtphone
 and dial a PSTN number using the integration which is not happening
 today.
As far as I know the FritzBox only supports dial through from SIP to 
PSTN after you enter a DTMF password at the dialtone as it considers all 
SIP connections to be trunks (i.e. telephone lines). Maybe you could 
manipulate the box with a software extension through FREETZ 
(http://www.freetz.org/) wich afaik includes a SIP server on the 
FritzBox. The site has an eglish section but be aware that the german 
site has more content.

Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FritzBox 7270

[Christian Victor]

2009/5/24 Manoj Panicker - FOES manoj.panic...@emirates.com

  Kare,
 Thanks much appreciated. It connected as soon as I created a SIP
 account. However I must try and figure out as how to get this box use IAX2.


Are you sure the FritzBox actually supports IAX2? As far as I remember it
does not.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY

[Christian Victor]

Duuh guys - it's so easy. Ever thought of simply compressing the compressed
data AGAIN???

Do that the necessary amount of times and - tadaa - it's done.

Chris

2009/4/1 Brent Davidson br...@texascountrytitle.com

  Cary Fitch wrote:

 It uses proprietary EDC.  (Extreme Data Compression)  The 140 bytes at 8
 bits each, and that is 2^140^8, a nearly inexhaustible key number which is
 related to audio and video data simultaneously stored on a Google Database,
 which is then sent to the user.

 Thus with the 140 byte message, full audio and video can be retrieved.

 This is an outgrowth of the data compression program circa about 1992, when
 disks were much smaller than today.  A very small compression program would
 infinitely compress data on a disk to allow storage of more data.  It was
 only a 200 bytes or so in size (DOS days):-) and worked perfectly.  Running
 it once resulted in lots of storage space.  It took very little time.  Of
 course rewriting the MBR (Master Boot Record) takes very little time.

 Recovering the compressed data was tough though.

 Cary Fitch
 04/01/09


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
 Sent: Wednesday, April 01, 2009 11:09 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL
 DRIVERFORASTERISK RELEASED TODAY

 On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote:


  I wish we could have this for real


  Micro-video-blogging: Limited to 140B ?




 I thought maybe it used Infinite Monkey Compression where a mathematic
 equation whose output over a specified domain would recreate the data-bits.
 For those unfamiliar with Infinite Monkey Compression it was theorized by me
 a few years ago as an offshoot of Infinite Monkey Theorem (monkeys,
 typewriters Shakespeare, etc...).  The original theory was that is an
 infinite number of monkeys could eventually type the complete works of
 Shakespeare through random coincidence then a random bit generator running
 for an infinite amount of time would eventually produce the equivalent bit
 sequence of any particular piece of software.  Infinity being, well, rather
 infinite and humans being mortal and all, infinite runs on a RBG didn't seem
 like all that great of an option, so I kept thinking...  Then I realized
 that any file can be represented by a sequence of numbers.  All you have to
 do is find the equation that will output those number sequences and you've
 got a highly-compressed way to recreate any file.  Just send the equation
 give it a start and end value and let the computer save the output as a
 binary file.  Unfortunately I was never able to take IMC beyond the purely
 theoretical.


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need to find small footprint asterisk platform

[Christian Victor]

2009/3/30 Peer Oliver Schmidt po...@theinternet.de

 The Horst-Box Professional has a lot of problems in the ADSL area
 (like stopping transfers after a dozen or so megabytes for example),
 and I have had lots of needs to hard-reboot the box, after enabling
 VoIP functionality.


Well - I never ran one in a professional enviroment and only use one single
unit in my home. Until now and with the latest firmware it runs without
bigger problems. But I agree that for professional use you neet to take a
close look at reliability.


 The D-Link support is useless (The answer the support request and
 without taking the answer into account close the ticket).


I agree - support is crap.It's basically here is the source so thats not
our probolem anymore.

The boxes would have been perfect for the german market, however, the
 way it was implemented, they are totally useless. And yes, I have
 tried all available firmware versions, and always made sure to follow
 the instructions to the letter, with regards to configuration reset.


You found instructions??? Lucky bastard! :-D

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need to find small footprint asterisk platform

[Christian Victor]

Here in germany D-Link sells a device called the Horst-Box 
Professional wich is a ADSL modem/router with WiFi and an integrated 
embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind 
serves me right. Size is about 180x250x50mm. Its been around for some 
years so maybe it is already EOL.

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card

[Christian Victor]

Andreas-Johann Ulvestad schrieb:
 When inserting the cable going into TE122 into an ISDN phone, the phone
 works perfectly.
   
That should not happen with an E1 line as your phone normally has a BRI 
(S0) connector with only two b-channels.

Seems that your line is configured ar BRI and not PRI. Either you got 
the wron wire or Telenor did something terribly wrong.

Hilsen ;)
Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP trunk with 250 lines

[Christian Victor]

Hi!

A customer of mine wants to connect an asterisk system with 240 to 480 lines
to a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.

Phones will be connected to the server through the same SIP trunk as this
will be some kind of a hosted pbx.

Given he finds a provider wich has this much SIP capacity and IP bandwith
and no codec conversion is needed - do you think this is possible with pure
asterisk on a decent system? Is there anything I shoudl watch out for?

Your help is much appreciated!

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP trunk with 250 lines

[Christian Victor]

2009/3/24 Cary Fitch ca...@usawide.net

  First Issue to be addressed is how many simultaneous calls and bandwidth
 availability.

 Number of “lines” (numbers) is not a limitation in it self unless they are
 all in use.

Sorry for being a bit unclear in this point. What I meant was 240 to 480
concurrent active calls.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP trunk with 250 lines

[Christian Victor]

2009/3/24 Danny Nicholas da...@debsinc.com

  Here are a few “look outs”;  Using conference rooms will increase your
 bandwidth requirements.  On board Network controllers will affect
 performance in this “high-use” scenario.  250 simultaneous calls will use
 about 7.5Mb of bandwidth depending on the codec(s) you use.

I think we will use G.711a to prevent transcoding. So I calculate with 25 to
50 Mbit. I hope a dedicated 1GB Ethernet (although on board) will do it.

MeetMe will not be used.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP trunk with 250 lines

[Christian Victor]

2009/3/24 Grygoriy Dobrovolskyy megaho...@gmail.com


 If the switch is fine why not ? But i wander why kind if switch is that
 240-480 fxo ? ;)
 Sounds like a big overkill.
 And i dont see a problem with asterisk, if not too much transcoding
 involved and with the right hardware.


 It's an ISDN telephony switch at a PSTN carrier. That type wich usually
connects T1/E1 lines to the telephony network but with SIP support built in.

Planned hardware is an AMD Opteron dualcore 2,3GHz with 4GB RAM.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP trunk with 250 lines

[Christian Victor]

2009/3/24 Danny Nicholas da...@debsinc.com

  I use a Dell with the 1Gb Ethernet on board, but had to clock it down to
 100 Mhz because * has an issue with Dell on board Ethernet.

Ah - good to know. I think we will use SUN machines. But I'll keep that in
mind.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] conference and wifi phones

[Christian Victor]

2009/3/24 Steve Gladden aster...@michiganbroadband.com

 I REALLY like the Snom M3 DECT SIP base.


Yeah - it's such a pitty that you always have to buy it bundled with one of
these crappy handsets. Or is there a way to get only the base that I don't
know?

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Magic SIP Phone

[Christian Victor]

Maybe the Siemens DE380 IP R could help you. It's a brand new IP phone with
an integrated router.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Magic SIP Phone

[Christian Victor]

  grandstream gxp-2000 works fine for that.
  depending on firmware rev its two ports are either a switch or router.

 Grandstream removed this functionality in recent softwware upgrades - I
 guess they needed the code space for other things.


Why would you want a router in the phone and not let the PC connected to the
phones internal (mini) switch get an IP from the DHCP server in the cable
box? Or will that deliver only ONE IP at a time?

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Portech MV3770 Caller-ID

[Christian Victor]

2009/3/11 Håkan Källberg h...@simulina.se


 Hello!

 Does anyone of you have Caller Presentation working in the other
 direction?? My mv370 is working well, execpt the Caller ID on outgoing
 GSM calls. This works with the SIM card/Provider I am using if I put
 the SIM card in a telephone, but not in mv370. I have tried options on the
 mobile setting page you talked about, with nor difference. I have
 tried to put the nummber of the SIM card as the Caller ID in the original
 Asterisk call too, just in case, but no.


All I had to do is to enable the Caller ID ind the Mobile-Settings dialog
for each SIM (something like presentation/revocation afair). I did NOT set
the GSM number anywhere nor do I send it from Asterisk.

Don't forget to save and reboot the gateway.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Portech MV3770 Caller-ID

[Christian Victor]

2009/3/10 Sasa s...@shoponweb.it

 Hi, I have modified in Mobile/Setting the parameter SIP From from
 tel/user to tel/tel and now I view the correct incoming number.
 Thanks.


Glad I could help. It took me nearly a month to figure that out. ;-)

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Portech MV3770 Caller-ID

[Christian Victor]

2009/3/9 Sasa s...@shoponweb.it

 Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2)  Portech MV-370,
 my problem is that when arrived an external call I don't view (on my
 internal phone) the phone number but I have the number extension that is

 ...


 ..now what parameter can I modify for to view the external phone number ?
 Thank in advance.


Hi Salvatore!

Can you verify if the number is submitted to Asterisk?

If not maybe you need to change the way the number is transmitted from the
gateway to the Asterisk box. I can't remember the exact parameter but it is
on the Mobile-Settings page. There should be four choices in a drop-down
list - various combination if SIP-ID/number/incoming number.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bounty- CDR Bug Fix

[Christian Victor]

2009/3/4 Atis Lezdins a...@iq-labs.net

 Bottle of Riga Black Balsam (45%), just have to figure out a way to send it
 :)


Balsam??? By mail? Doesn't that count as liquid explosive? ;-)

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] call file concurrency

[Christian Victor]

2009/2/27 Bill Michaelson b...@cosi.com

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?


Afaik only by limiting the number of call files in the directory.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] GSM codec is a good choice ???

[Christian Victor]

2009/2/25 Alejandro Cabrera Obed aco1...@gmail.com

 But in my case, I don't need trascoding because every chanel is in GSM
 and voicemail has gsm sound files.

 And for the moment, my Asterisk is not connected to the PSTN, so there
 is no trascoding gsm-to-PCM or to analog.

 So I think gsm is a good choice for my scenario, do you ???


Hi Alejandro!

Just to answer your question clearly: Yes, GSM would be a working option for
your scenario.

If you ever need G711 to connect to ISDN the transcoding should be no
problem for a P4 class system and for example 30 ISDN-Lines.

But what the others want to say is that buying new phones just to avoid
paying for G729 licenses may not be a good idea as the licenses are quite
cheap (US$10 for every transcoding you USE at the same time, NOT for every
phone you have).

I hope that answers your question.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Quiet 24 port POE gig switch

[Christian Victor]

2009/2/2 Singer XJ Wang w...@pythian.com

 [snipped]

 You can do that by using fans other than the tiny, whiney, 40mm fans
 that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
 fans at the back or front, pushing air in (hence the deep
 dimensions), but the top and bottom would need recesses to allow
 sufficient airflow when the positions above and below are filled.


 How are you getting these 80 or 120mm fans in a 1U chassis? Remember you
 got barely 45mm to play
 with at the back and front of the switch. How are you going to mount a 80mm
 or 120mm fan on there? Are you assuming that the units mounted
 above (or below) your switch is a short 1U? You can't assume that...


Ever heared of a centrifugal fan? ;-)

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Attacking DECT

[Christian Victor]

2009/1/1 Olivier oza-4...@myamail.com


 To attack DECT equipments, a ComOnAir module was used.
 This module is a PCMCIA addon which provides DECT connectivity.
 I don't think this module is available or manufactured anymore.
 So it seems difficult for anyone to reproduce this DECT attack.


Or you could buy it in numbers on eBay.de for € 19,99 ;-)

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] whisper time remaining

[Victor Alvarez]

Morning! Thank you very much for the answers. These are my considerations:

 If it's a pre-paid app and you're doing a Dial command (like a calling 
card), why not use the limit (L) feature that's built in?

Because limit the time before the call starts does not cover all the 
possible cases of use of a prepaid application, like a situation in 
which the user is consuming credit in more than one way. And also 
because whisper i.e 'you have one minute remaining' is a useful 
information for the user and a very common requirement for these 
applications.

  If you know the channel that you need to ‘whisper to’, You could 
always create a call via the manager to the whisper application and 
bridge it to PlayBack of a gsm file that will be played or send it to a 
context that will announce the time left. Then drop off….


I know the channel that I need to whisper to. This idea sounds certainly 
more close to what I had in mind, I will need to translate it into code.

  From a technology standpoint, if I'm not mistaken, this would require 
some sort of conferencing because it is mixing two digital audio 
streams. Call it what you like, but it has to have extra resources.

It depends on how you bridge the channels. It is clear is that, from a 
programmer point of view, I would have preferred to call it it an 
Asterisk Manager command :-)

Kind Regards,
Victor

Alexander Lopez wrote:

 If you know the channel that you need to ‘whisper to’, You could 
 always create a call via the manager to the whisper application and 
 bridge it to PlayBack of a gsm file that will be played or send it to 
 a context that will announce the time left. Then drop off….

 Alex

 P* **Kindly consider the environment before printing this e-mail**.*

 

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Fred 
 Posner
 *Sent:* Monday, October 27, 2008 10:16 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] whisper time remaining

 On Oct 27, 2008, at 7:31 PM, Victor Alvarez wrote:

 Hello everyone,

 I'm trying to find out a way to whisper the time remaining for a

 prepaid application on a established channel. Unfortunately I think

 there is a lack of PlayBack/Background commands which can be 
 applied on

 a working channel as well as a lack of spy/whispering commands 
 available

 via Asterisk Manager. Does anyone know how to implement this?

 Thanks a lot.

 Regards,

 Victor

 If it's a pre-paid app and you're doing a Dial command (like a calling 
 card), why not use the limit (L) feature that's built in?



 Fred Posner

 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]*

 Tel: +1 (212) 937-7844 x501

 Fax: +1 (954) 252-4187

 www.teamforrest.com http://www.teamforrest.com

 

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] whisper time remaining

[Victor Alvarez]

Hello everyone,

 I'm trying to find out a way to whisper the time remaining for a 
prepaid application on a established channel. Unfortunately I think 
there is a lack of PlayBack/Background commands which can be applied on 
a working channel as well as a lack of spy/whispering commands available 
via Asterisk Manager. Does anyone know how to implement this?

 Thanks a lot.

Regards,
  Victor

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] whisper time remaining

[Victor Alvarez]

 Thanks for the prompt answer, Alex. Considering whispering is a basic 
functionality for an IVR application, I think there should be a better 
way of doing this, not implying the use of meetme.

Alex Balashov wrote:
 The only way I can think of is if you use conference rooms to 
 cross-connect the call.

 Victor Alvarez wrote:

   
 Hello everyone,

  I'm trying to find out a way to whisper the time remaining for a 
 prepaid application on a established channel. Unfortunately I think 
 there is a lack of PlayBack/Background commands which can be applied on 
 a working channel as well as a lack of spy/whispering commands available 
 via Asterisk Manager. Does anyone know how to implement this?

  Thanks a lot.

 Regards,
   Victor

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 


   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call files

[Christian Victor]

Steven Howes schrieb:
 Have created a system that involves using call files in the outgoing  
 spool folder. On some occasions it retries which is fine is there  
 any way to view calls waiting retries from the CLI? Using 1.4 btw.  
 Have googled to no avail (although it is near the end of the day so I  
 might be being a muppet!)

One solution would be to look INTO the callfiles. The content of the
file changes if there is a retry involved. I don't know by heard how
exactly it changes but afair there is a line stating the retry time.

Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PoE switch recommendations?

[Christian Victor]

Hi Ken,

we are quite satisfied with Linksys SRW248G4P. 48 port PoE, 4 GB uplinks
and 2 GBIC slots. VLAN, QoS and all the like is on board. Around US$600
I guess.

Only drawback in my opinion is that they are loud like a starting
airplane. You definately don't want them next to your desk. ;-)

Christian

Ken D'Ambrosio schrieb:
 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the cash and
 do a Cisco system, but I'm relatively certain that would get shot down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
 so long as it has Gbit uplinks.
 
 Thanks!
 
 -Ken
 
 
 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 


-- 
victormedia
jahnstraße 105
40215 düsseldorf
germany

fon +49 211 5833434
fax +49 211 5833435
sip [EMAIL PROTECTED]

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Pressing 0 to get an external line

[Christian Victor]

Hi Asterisk users!

I have a little problem with an Asterisk 1.4.22 installation for a
customer. The PBX is connected to an E1 line and we have a few snom 300
attached to it.

The goal is to emulate traditional german PBX behaviour wich is the play
a stuttered internal dialtone after pickup and accept all internal
extensions. If a specific get office digit - usually 0 - is pressed
you get the normal dialtone. Now you can enter the external telephone
number end get connected immediately without pressing dial on the snom
phone.

My problem here is first to implement this internal/external dialtone
and second to make the phone dial without pressing dial on the phone.
Because the length of telephone numbers in germany is not constant we
will have to implement this with a timeout or by somehow overlap-dialing
digit by tigit to the E1.

I hope one of you already implemented this behaviour or can maybe push
me in the right direction.

Thanks a lot
Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Pri to sip interfaces

[Christian Victor]

Tom Moore schrieb:
 What are your suggestions to people who have pbx systems that interface with
 the world over pri and want to convert them to sip interfaces so that they
 can use sip trunking?

I'd go for a Patton SmartNode. See www.patton.com - they have SIP
gateways up to 4 T1/E1.

Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI Splitter

[Christian Victor]

2008/8/31 Olivier [EMAIL PROTECTED]


 What happens if the PC supporting this card is powered off ?


It is powered over USB from the main (internal USB) and backup (external
USB) server. If one of the power fails it will switch to the other server.
If both servers power fail you have a problem anyway. ;-)

Do you have an idea of its price ?


Approx. US$ 700

Regards
Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI Splitter

[Christian Victor]

2008/9/1 Karl Fife [EMAIL PROTECTED]

  It is powered over USB from the main (internal USB) and backup (external
  USB) server. If one of the power fails it will switch to the other
  server.
  If both servers power fail you have a problem anyway. ;-)

 This is incorrect.  According to Jim Rhodes at Rhino, there is no NEED
 for 'backup' power from another server via USB:


Did I write you NEED a second power supply? I was just refering to the fact
that when both servers power fail you have a problem no matter if the
failover switch ist still working or not.


   Do you have an idea of its price ?
 
  Approx. US$ 700
 

 This is incorrect.  Again, according to Jim Rhodes, the FULL
 nobody-ever-actually-pays-this-much LIST price is US $350.  In my
 estimation, the street price will be between US $220 and $299 depending

on your reseller's markup.  I don't know if Rhino has M.A.P. rules.


I hope you are right. Maybe this guy should share his information with the
world. According to
http://store.variantdistribution.com/category-s/49.htmVariant - one of
Rhinos distributors and the only source I was able to find
- quotes the card for US$ 700.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IVR question

[Christian Victor]

 I'm setting up my IVR system, how can I register in a mysql database the 
 IVR menus accessed by the clients ?

Just use the MYSQL-Functions in the dialplan to write the menues name
(and datetime maybe) in a table.

To access MYSQL from the dialplan you need to have the asterisk-addons.

Christian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BRI AND DATA connection

[Christian Victor]

Anton schrieb:

 Does anyone tried BRI with asterisk for DATA transfer? My 
 customer
 wants BRI connection, but he wants it for the data, and I 
 have to
 bring connection to his office, so I see the connection as 
 follows:
 
 E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so 

Why would someone want to put Asterisk and IAX in the data path?

I would do E1 routedd to  TCP/IP routed to BRI

Chris

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Website callback

[Christian Victor]

I don't know if there is something like that prebuilt. But is seems to be
quite easy. Push the call events in the database, let a cron run ever minute
and create a .call file for evry call thet is due.

The alternative is to not use a database and create a .call file with a
future date/time. Afaik asterisk processes only callfiles with a past
date/time.

In the call context ask the callee to press a digit to be sure he is human
(press one to be connected to one of our agents) and then - as you said -
drop him in a cue.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 1.4.20.1 problems

[Christian Victor]

Hi!

I just upgraded my Asterisk server from 1.4.6 to 1.4.21 and now I experience
some strange behaviour.

1) The Asterisk CLI (asterisk -r) stops responding after some minutes. I
cant CTRL-C or exit the CLI anymore and no activity is shown. Just like if
the connection is interrupted.

2) When I dial into the machine via ISDN (Sangoma A102dx) everything works
okay. But when I hang up it takes about 10 seconds until asterisk recognises
the hangup and jumps to the h extension. pri intense debug shows the
DISCONNECT message immediately after I hangup the phone.


Did anybody of you encounter similar problems or do you know what the
reason/solution could be?


Thanks a lot
Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] CLIR missing in MySQL CDR records

[Christian Victor]

Hello!

I just encountered a strange thing in my mysql cdr records. From a
certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR
flieds in the cdr table. As far as I know no changes happened to the
system on that date and until then CLIR are recorded properly.

The CLIR is still transmitted by the PRI and is shown in the console
when a call comes in. But no traces of it in the CDR.

Did anybody of you ever experienced this?

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to Configure 1.4.17 to Store CDR's in PostgreSQL

[Victor]

I'm having a heck of a time saving my CDR's into a PostgreSQL database. I've
installed PostgreSQL on a remote server and it is successfully storing voicemail
messages but I cannot get the 1.4.17 system to store CDR records there.

Has anyone successfully configured a 1.4 system to store CDR's in a remote
PostgreSQL database?

If yes, can you point me to any set of instructions that work. I've tried the
instructions outlined in the Asterisk TFOT 2nd edition book as well as
voip-info.org to no avail.

Thanks in advance,

Vic


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO

[Victor Toofic]

El Sat, Jan 19 de 2008 a las 23:35 -0600, Moises Silva comentaba:
 First, let me say I am confused about this:
 
  I've changed the line (chan_unicall.c):
 
  uc_callparm_calling_party_category(callparms,
  UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
 
  to
 
  uc_callparm_calling_party_category(callparms,
  UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
 
  because without this I cant receive calls from the telco. With or without 
  this I
  can't place calls to the pbx.
 
 I am quite sure you have made a mistake in this statement, why? simply
 because this code is executed when YOU START the call to the far end
 (whatever it is, Telmex or the other PBX), so it makes no sense to say
 that w/o that change you can't receive calls, no sense at all. I am
 sure you messed up somewhere else in the configuration files just like
 possibly you are doing right now for the PBX.

Hmmm.. I was pretty sure that if I dont change that line I cant receive
calls from telmex.. but let try it againg and I will tell you what it was.

 In anycase, I am about
 to make a new release of chan_unicall Asterisk driver that will
 include a way to modify the calling party category from the dialplan
 extensions.conf

Wouldn't it be better if that could be done in unicall.conf? As with the
other options like protocolvariant and protocolend ??

Anyway.. thanks for doing that update ;) I would be glad to know when it
is available.

 Now, regarding your problem when receiving calls from the pbx, I think
 you have configured the PBX to not send ANI digits, and you configured
 chan_unicall to expect ANI digits, hence the timeout. Try configuring
 Asterisk with 0 callerid for the PBX side, or configure the other PBX
 to send the proper number of ANI digits.

Well.. in the first place.. that pbx is not mine, I didnt configured it
and I cant even touch it, Im just putting asterisk in between right now.

Im gonna try that.. Thanks for the help!

--
Regards..
Victor Toofic

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users