Re: [asterisk-users] tls on asterisk 13

2016-03-19 Thread Hans Witvliet
On Wed, 2015-07-08 at 15:09 -0400, Ryan, Travis wrote:
> Asterisk13 can do native tls with each phone? Nice.

Some soft phone support TLS,
but does anybody knows a soft phone that support pkcs11?
(keys & certs stored on a smart-card)

Hans

-- 
_
-- 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] Asterisk Phone ( Telecom feature )

2014-10-07 Thread Hans Witvliet
On Tue, 2014-10-07 at 08:37 -0500, Don Kelly wrote:
 JG confirmed that it is possible, but it has not been defined.
 
 Without knowing what kind of instruments you are using, a possible it
 would be for a party to dial a 4-digit extension number to talk to someone
 internally, completing a call without using the PRI trunks.

Indeed, it is rather vague.
The intercom I came across the last couple of decades, were simple
analogue-phones, without a dial-pad, with only one button.
If you lift the receiver and press that single button, you'll ring the
other phone. And vice-versa.

That sounds do-able, for any kind of phone connected to Asterisk.
With a wildcard in the dialplan, you can create an truly any-key :-)

But the O.P. might expect something else

-- 
_
-- 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] Does Asterisk 1.8. Supports Video Calls

2014-09-08 Thread Hans Witvliet
On Tue, 2014-09-02 at 13:18 -0500, Khalid Touati wrote:
 so it seems Asterisk Versions does not support video I guess
 
 
Used it with jitsi and linphone softphones, works just OK.

Just for testing i did a video-call on the loop-back, great test tool
for showing the influence of (limited-) bandwith / latency.

Ideal for demo's

Hans


 

-- 
_
-- 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] Attack on Sip server.

2014-07-01 Thread Hans Witvliet
On Fri, 2014-06-27 at 22:24 +0530, Anurag Rana wrote:

 
 iptables -I INPUT 1 -p tcp --dport 5060 -m string 
 --string VaxSIPUserAgent --algo bm -j DROP
 
 
You make a fundamental mistake here.
Firewalls (both inline and hostbased) should drop everything by default.
And you should specifically accept what you are expecting and capable of
handling. Not the other way round.

Above rule is something like:
The front door is locked between 9:30 AM and 10:15 AM, as you expect
burgers to come to your house.


-- 
_
-- 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] High Availability with Asterisk

2014-03-09 Thread Hans Witvliet
On Sat, 2014-03-08 at 20:27 +, ad...@3a.hu wrote:
 My approach (in theory only, so please correct me if I'm wrong) would be 
 to run asterisk on multiple boxes (one each).  A dedicated monitoring 
 box (nagios?  custom scripts?) would perform frequent checks against the 
 boxes (one of my previous projects one asterisk was using call files to 
 demonstrate its health to another one).
 
 If a box fails, I would simply redirect/reroute its traffic to another 
 one, using network solutions.  Such as shutting down the production 
 interface of a suspectedly failed asterisk box, having an idle one pick 
 up its IP address, or using load balancing / routing / NAT to redirect 
 the client's traffic to a standby box.
 
 My approach is based on the experience that linux based HA tools are 
 often not free, or don't scale well, or engineered to circumvent an 
 error in a slower manner (eg. booting a second VM takes too much time). 
   However in the network world, there are well known protocols that were 
 designed to take over in a matter of miliseconds.
 
 I do understand that this would not provide 'session' data, so failing 
 over to a different box would mean the need to re-register, could cause 
 calls to drop etc.  This might be unacceptable for you.  As I said in 
 the beginning, I haven't been building such systems, in my experience a 
 dropped call is not that big of a deal, if it happens because the 
 network cuts over to a different box.  This could be handled with a pair 
 of frontend load balancers, where the number of asterisk boxes can be 
 transparent.
 
 hope this helps
 adam
===

Hi Adam,

Don't confuse high availability with load balancing, as these two
are not related. These two have totally different objectives and are
achieved in different ways.
Either/both of them can very well be achieved with opensource tools.

Even with commercial software is maintaining call when a intermediate
PABX breaks down nearly impossible




-- 
_
-- 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] Lots of calls, less memory

2014-02-10 Thread Hans Witvliet
On Mon, 2014-02-10 at 10:39 -0500, Tech Support wrote:
 Rather than speculate, take a look at the output of top. If you're
 running out of memory, shut down useless processes. You'd be surprised what
 processes get started by default that you don't need. You should also check
 the Asterisk logs and look at the last few things Asterisk did right before
 it restarted. You may also want to consider not loading Asterisk modules
 that you are never going to use. Just a suggestion.
 Regards;
 John
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
 Sherrill
 Sent: Monday, February 10, 2014 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Lots of calls, less memory
 
 On 14-02-10 9:46 AM, Mike wrote:
  What log entries are leading you to think that you're running out of RAM?
 
 None.  It's just my guess.  The log doesn't show anything except Asterisk
 restarting.

how about running free or vmstat inside cron every hour or so?
If you do a vmstat 1 10 each hour on the hour, it tells you 10 times
with one second interval the amount of mem you got.
If you do that within cron, you can see the difference during a couple
of days.

Running out of mem, will cause unexpected results (to be found in
syslog), though rebooting should not be one of them. 

for unintended reboots there are a lot of hardware related causes though
Some are easier to detect (like high temp) some are harder.
My most favorite is a moron-co-worker, touching sensative parts (cpu,
mem, mobo) with his ESD-unprotected hands. Problems might show up even
after months or years after the crime has been commited.

hw





-- 
_
-- 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] Multiple IAX2 Trunks Load balancing

2013-12-13 Thread Hans Witvliet
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
 On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
  Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want 
  to load balance incoming calls over IAX2 trunks. If any trunk goes 
  down the calls traffic will be shared with other available trunks.
  When it gets Up the script is supposed to perform as desired i.e in 
  load balance mode.
 
  Thanks in advance.
  
 
 Hans said:

 
 Perhaps it is possible to do the L.B. at the O.S. or network level, and let
 all trunks appear to asterisk to one single trunk.
 
 Don asks:
 
 What's the value of load balancing multiple IAX trunks between the same
 system pair? What resources are being balanced?
 
++

Perhaps the O.P. can explain about his intentions...

In some situations it makes sense though:
If you have to connect two servers, and use different kind of
infrastructure / multiple providers...

hw


-- 
_
-- 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] ISDN outgoing caller id

2013-08-28 Thread Hans Witvliet
-Original Message-
From: Gergo Csibra csi...@gmail.com
Reply-to: Gergo Csibra csi...@gmail.com, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ISDN outgoing caller id
Date: Tue, 27 Aug 2013 21:28:36 +0200

Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:

 On 08/27/2013 08:04 PM, Gergo Csibra wrote:
 Hi,

 is anybody out there who can set the outgoing caller id on ISDN (CAPI
 or misdn) channels? I've tryed everything what I found in forums, os
 voip-info.com but no luck. I use a fritz card with CAPI in my first
 installation (1 BRI), and a hfc 4 port bri card with misdn on other.
 The first installation have p-t-mp configuration, the second one is
 p-t-p. Both configuration is EuroISDN in Hungary.

 So, can anybody help me?

 Have you checked with your Telco if they allow you to change the 
 callerid? If yes, are you setting the callerid to a number that you are 
 allowed to use? You can't just set callerid to any number you like. You 
 must own the number which you want to set callerid to. I have no 
 problem setting the callerid on outgoing calls via chan_capi to one of 
 the numbers that the telco assigned to me.

Yes, of course I want to set our assigned numbers, becuse the called
party sees Unknown now.
-Original Message-
It's been a while ago for me, but:

Besides the item mentioned above (hit that one also) two things come to mind..
1) is CLI-Display activated on that line? For some telco's it is a fascility 
that has to be enabled..
(you can check it by plugging in a isdn-handset, and try to make a call)

2) Perhaps accidentally activated the HIDE CLI activated?


hw

--
_
-- 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] Performance Asterisk large installation on Vmware/Xen

2013-05-20 Thread Hans Witvliet
-Original Message-
From: Rafael dos Santos Saraiva rafaels...@gmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Performance Asterisk large installation on
Vmware/Xen
Date: Sat, 18 May 2013 15:01:06 -0300

Hi


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


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

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



--
_
-- 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] НА: asterisk-users Digest, Vol 105, Issue 40

2013-05-07 Thread Hans Witvliet
From: virus.c...@mail.ru virus.c...@mail.ru
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40
Date: Tue, 07 May 2013 07:53:53 +0600

help

-Original Message-

exten = 911,1,Answer()


--
_
-- 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] looking for a way to do appointment reminders

2013-04-26 Thread Hans Witvliet
-Original Message-
From: jg webaccou...@jgoettgens.de
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] looking for a way to do appointment
reminders
Date: Fri, 26 Apr 2013 09:33:42 +0200

Hi Brandon!

I have a wakeup call system based on call files that are generated 
by an external C program. The call files can be triggered by dialing a 
phone number (e.g. for waking up the hotel guest in room 333 at 6:15 am: 
*77*3330615) or from outside via a web interface, or whatever.

It looks like your task has the same basic requirements. Setting up a 
call file based system is not very difficult, but details like 
pronunciation of the guest's or patient's name may involve some 
additional work.

-Original Message-
I dare to disagree.

Phone call as a reminder won't work.
When would you call them? An hour in advance?
They will probably never make it in time, or you disturb them at an
inconveniant moment.

Only reasonable option is to send them an SMS.

hw


--
_
-- 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] Asterisk Calendar integration suggestions

2013-04-25 Thread Hans Witvliet
Might have a look at tine:
http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration

hw

-Original Message-
From: Steve Totaro stot...@totarotechnologies.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Calendar integration suggestions
Date: Thu, 25 Apr 2013 10:43:52 -0400

Without knowing requirements, Sugar CRM seems to be the most supported.
 


Thanks,
Steve Totaro


On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us j...@millican.us
wrote:
Hello all,
I am looking into building a calendar server (due to business
requierments I can not use public hosted calender like Google),
and am looking for suggestions based on experience with
different calendar applications/servers available for Linux that
you have integrated with Asterisk.  If you can give a quick,
simple list of what worked and what didn't I would be very
grateful.
Thank You,
John


--
_
-- 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



--
_
-- 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] Installing Asterisk on Virtual Machine

2013-04-23 Thread Hans Witvliet
Could it be distro-related?

I have various versions of asterisk (from 1.4 upto 11.3) running
paravirtualized or HW-virtualized with XEN.
Normally i use the pre-build packages from suse, only when i want to try
a release-candidates i need them myself.

hw

-Original Message-
From: Sandeep Raju sandeepr...@practo.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Installing Asterisk on Virtual Machine
Date: Tue, 23 Apr 2013 10:18:00 +0530

Hi Tzafrir,


I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance
running on my private openstack cloud. My bare machine is Intel® Core™
i7-2600K CPU @ 3.40GHz × 8  with 8GB ram and running at 64 bit ubuntu
12.04 desktop edition with Kernel Linux 3.2.0-23-generic. 


output of uname -a on my ubuntu cloud instance where i'm trying to setup
asterisk..

Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC
2012 x86_64 x86_64 x86_64 GNU/Linux



Here is my backtrace.. http://paste.kde.org/730316/


Sorry for the late reply...


On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote:
 Hi,

 I'm trying to install Asterisk 11.2 on a virtual machine in my
private
 opestack cloud.. When I compile Asterisk 11.2 from source
(./configure,
 make, make install) as specified in the Asterisk book and run
it, it gives
 me the error: Illegal instruction (core dumped).

 Any ideas how I can solve this?


What operating system do you have installed there? What CPU?

What is the output of:  uname -a

Illegal instruction means that you tried running an instruction
that the
CPU cann't run. Maybe an incorrect choice of optimization flags?
Maybe
this is due to libraries not matching your architecture?

Next thing to do: get a trace from the core file that was dumped
using
gdb.

--
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

--
_
-- 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



--
_
-- 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] IPv6

2013-03-29 Thread Hans Witvliet
Hi all,

I had to re-install a new machine and noticed that by default, ip was
only listening on 0.0.0.0, thus ipv4 only. Easily changed.

However, when looking at iax.conf, I found here the same, but it looks
like iax is still ipv4 only?
If i change bindaddr=192.168.0.1 towards bindaddr=::, and look with
lsof -i iax is still not listening on V6.

Is iax/ipv6 still on the TODO-list ?


Hans



--
_
-- 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] chan_mobile: FXS

2013-03-27 Thread Hans Witvliet
Hi all,

Finally i got hold of some bt-dongles that seems p[retty stable, the
asus-bt211.

After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres
and blackberry address) and mobile show devices is showing me that the
BT-link is up, and remains stable up.

Seems good, but it looks like asterisk is seeing the BB as a trunk/FXO.
However, i want to use the phone as an FXS.

Before ending up in trying something that was never foreseen and perhaps
even impossible, i was hoping that i could use the BB as an oridinary
audio device and still use the keys on the phone for starting/ending
calls, and the dialpad for selecting phone numbers.
And having the connections go (via BT) through asterisk instead of GSM.

Is this possible at all, or am i embarking on a mission impossible ;-)


Hans




--
_
-- 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] SIP account registration fails after upgrade to 1.8

2013-03-22 Thread Hans Witvliet
-Original Message-
From: Jaap Winius jwin...@umrk.nl
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP account registration fails after
upgrade to 1.8
Date: Fri, 22 Mar 2013 02:46:43 + (UTC)

On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:

 Hopefully, my ISP will see fit to squash this bug ASAP.

Well, I got my answer from them quickly enough: Nope.

Luckily, somebody was kind enough to suggest a workaround. Unfortunately, 
it involves, downloading the source code and making a few changes to it 
to prevent Asterisk from adding '@IPaddress' to the end of the Call-ID 
string. Nevertheless, it's easy enough to do. The idea is to look for 
this string that appears twice in ./channels/chan_sip.c:

  ast_string_field_build(pvt, callid, %s@%s,
  generate_random_string(buf, sizeof(buf)), host);

And to change it to:

  ast_string_field_build(pvt, callid, %s,
  generate_random_string(buf, sizeof(buf)));

Now my Call-IDs look like this:

   Call-ID: 63935a8d2144d4f1309024fd7612f608

Instead of this:

   Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a]

Still, I'd much prefer that my ISP fixed the problem instead, because now 
every time a security update becomes available for Asterisk, I'm going to 
have to download the source code, make the same changes, recompile it and 
install it all over again and again. Ho hum.

Of course, an even better solution would be if Asterisk had a variable 
with which to alter the Call-ID string format so that I could omit the IP 
address. :-)

Cheers,

Jaap

-Original Message-

Hi Jaap,

just wondering, might this perhaps be an IPv6 quirk?
By altering '@IPaddress'  you got rid of : '@[2001:888:abcd:1::a]'
Does the dame happen with V4-only?

I presume you didn't activate V6 at your end lately?
Other idea (perhaps pointless), you got the numeric address, would the
same issue still exists if '2001:888:abcd:1::a' could be translater back
into a dns-name? (include it in your /etc/hosts ?)

Sometimes the '[]' cause some side-effects (specially if some regex are
used unseen)


Groet, hw


--
_
-- 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] chan_mobile

2013-03-12 Thread Hans Witvliet
-Original Message-
From: Emiliano Vazquez emilianovazq...@gmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_mobile
Date: Tue, 12 Mar 2013 18:01:34 -0300

El 10/03/13 13:18, Hans Witvliet escribió:
 Hi,

 I've been looking at the list at:
 http://www.voip-info.org/wiki/view/chan_mobile

 But when googling of any of the known working devices, there ain't any
 for sale anymore, probably replaced by more recent types.

 So, anyone around here who bought recently an BT-dongle that is working
 with asterisk?

 hw
Hi Hans!

You can try with chan_dongle [1]. Whit this you can use USB Modems from 
Huawei to get a full trunk running with your asterisk box.
It's nice to use, can send and receive sms from your company. The 
quality is good and really better than chan_mobile.

Best regards.


[1] http://wiki.e1550.mobi/doku.php

-Original Message-

Hi Emiliano,

thanks for your reply,
I think i might use it for a different project, I got an huawei-E1820


But at the moment i have to look at something else:

The issue is contacting people not currently in the office.
I've been trying to accomplish secure voice with a softphone through a
vpn-tunnel, but the choosen softphone turns out less reliable then
expected.

While still working on that thread, other option is to equip each laptop
with a _proper_ blue tooth dongle, and use their dumb/smart phone as an
USB-audio device. If they are near their laptop, presence should allow
me to use chan_mobile. (with an additional advantage not having to pay
GSM-providers abroad)

So, main issue is stability, reliability and usability for end users.

Unless i can use a huawei as a single-channel BTS, i'll have to stick to
use a BT-dongle.

Hans



--
_
-- 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 card and virualbox

2013-03-11 Thread Hans Witvliet
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] digium card and virualbox
Date: Sun, 10 Mar 2013 20:18:52 -0700 (PDT)

I am not mixing. I need this for LAB testing. 
How? This PCI passthrough, how to enable it on virualbox?
---
  Hi All;
 
  How to let the virualbox (ubuntu OS) to be able to see
 the digium card? Because when I install elastix or asterisk
 with dahdi, it is not able to see the digium card if the
 installation though the virualbox .. What is the solution?
 The solution is to run Ubuntu and Asterisk on your hardware
 natively, 
 not through VirtualBox.
 
 Virtualisation and high-performance hardware such as
 telephony cards  
 (it will be creating 8000 interrupts per phone line per
 second)  do not 
 mix, I am afraid.

What you might do, is  running an very elementary asterisk on the iron,
just acting as an PSTN-gateway. 
And run your experimental asterisk as a virtual client.

HW


--
_
-- 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] chan_mobile

2013-03-10 Thread Hans Witvliet
Hi,

I've been looking at the list at:
http://www.voip-info.org/wiki/view/chan_mobile

But when googling of any of the known working devices, there ain't any
for sale anymore, probably replaced by more recent types.

So, anyone around here who bought recently an BT-dongle that is working
with asterisk?

hw


--
_
-- 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] asterisk with 1000 extensions

2013-03-08 Thread Hans Witvliet
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk with 1000 extensions
Date: Thu, 7 Mar 2013 09:30:31 -0700

On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
 This is not school assignment or home work :)  We need to
 setup in society buildings. Each flat will have SIP extension
 (hard phone) registered on asterisk server. Calling
 between SIP extensions is required. No PSTN / ITSP SIP
 trunking. Just like inter-com feature.
  
 One way is to install 1000 IP Phones one at each flat
 Secondly, install multiple-line SIP gateways with RJ-11
 cabling.
  
 Is there any other low budget solution for this setup?
 


Grandstream makes some inexpensive phones that are still very good.


Cheapest hasn't been defined yet.  What's the budget?  Is there
existing networking at these locations?  Will you need switches?  PoE?

-Original Message-

I think Carlos said it properly.
Anything related to asterisk is insignificant compared to the rest.

I dare to say, that the requirements if for 1000 people to communicate
between themselves.

So why SIP-phones? Why VOIP at all?

Look at it a bit broader: network, maintenance (people), power, ...





--
_
-- 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] Error to install Asterisk

2013-03-05 Thread Hans Witvliet
-Original Message-
From: termo termosel fermit...@hotmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk
Date: Tue, 5 Mar 2013 14:30:05 +

Hi,


if I write du -sh the response is 271M. I don't know that it means.


Thanks,
Jordi



-Original Message-

Hi Jordi,

The du utility will show you the Disk Utilisation (hence the
abbriviation du)

What might be more relevant, is how much space is free.
That you can examine with: df -h


hw


--
_
-- 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] GSM Sip Gateway

2013-02-24 Thread Hans Witvliet
Are these 4G comaptible 


-Original Message-
From: Frank fr...@efirehouse.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] GSM Sip Gateway
Date: Sun, 24 Feb 2013 07:40:19 -0500

USA, this will be use with a 4G network. 

On Feb 24, 2013, at 5:24 AM, longst longst...@gmail.com wrote:

 where are you from by the way 
 
 Sent from Shitian Long
 
 
 On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote:
 
 Hi all,
 
 Anyone ever used GoIP GSM SIP Gateways ?
 If yes, what was your experience with those ?
 
 I'm looking at this:
 http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c
 
 If anyone has any (good) experience with another brand, I'll take the names 
 and models.
 
 Thanks
 
 --
 _
 -- 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

--
_
-- 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] GSM Sip Gateway

2013-02-24 Thread Hans Witvliet
Thought so.

make me wonder, as the O.P. specifically mentioned 4G.
But all the functionalities offered by 4G hardly seems to be relevant
for an asterisk-GW. Not?

If only for speech, first generation should be enough ;-)

hw

-Original Message-
From: John Novack jnov...@stromberg-carlson.org
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] GSM Sip Gateway
Date: Sun, 24 Feb 2013 09:15:37 -0500

From the Freq. list given on eBay, I don't think they are. The listed
freqs. are worldwide GSM since the mid 90's, but not 4G

John Novack

Hans Witvliet wrote:

 Are these 4G comaptible 
 
 
 -Original Message-
 From: Frank fr...@efirehouse.com
 Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] GSM Sip Gateway
 Date: Sun, 24 Feb 2013 07:40:19 -0500
 
 USA, this will be use with a 4G network. 
 
 On Feb 24, 2013, at 5:24 AM, longst longst...@gmail.com wrote:
 
  where are you from by the way 
  
  Sent from Shitian Long
  
  
  On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote:
  
   Hi all,
   
   Anyone ever used GoIP GSM SIP Gateways ?
   If yes, what was your experience with those ?
   
   I'm looking at this:
   http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c
   
   If anyone has any (good) experience with another brand, I'll take the 
   names and models.


--
_
-- 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] Asterisk SMS()

2013-02-19 Thread Hans Witvliet
-Original Message-
From: A J Stiles asterisk_l...@earthshod.co.uk
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk SMS()
Date: Tue, 19 Feb 2013 16:50:10 +

On Tuesday 19 February 2013, Nicholas Johnson wrote:
 Thanks for the help.  Right now I'm running asterisk on a raspberry pi
 using a phone number from flowroute.  Is using a company like flowroute
 the same as connecting to the PSTN?  Also i've tried to install smsq but I
 couldn't find any good documentation to get it setup properly.  So no, I'm
 not using smsq.

The bad news:  You need a GSM modem to send SMS messages.

The good news:  It is not so.

You can send SMS messages on POTS or ISDN lines
See the voip-wiki about it

--
_
-- 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] Asterisk calls between 2 private networks

2013-02-07 Thread Hans Witvliet
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
Date: Thu, 7 Feb 2013 10:36:36 -0700

On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote:
AJS,

That is a solution that I am envisaging.
But I would really love to try to work out with my issue first.
It will allow me to deploy more phones in separates buildlings
in the future. If I do the IAX solution, it means that for every
building, I need a box.. Which I would like to prevent.


Adding more points of failure and more devices to maintain without any
real benefit is always the wrong thing to do.  IAX is also flaky as
hell.


-- 

_

Carlos, 

with regards to your comment about IAX, where can i find your
bug-report?

--
_
-- 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] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

2013-02-01 Thread Hans Witvliet
-Original Message-
From: Olivier oza_4...@yahoo.fr
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote
LAN workstation
Date: Thu, 31 Jan 2013 08:25:42 +0100

Hello,

On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified) and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but I'm not
certain I'm looking at the right direction.

Regards
--
_
Hi Oliver,

I've been trying to do this for a while.
Been using latest blackberries and oldest nokia, and a laptop with
build-in and also an external BT-dongle.

What i noticed, is that the presence is highly unstable.
Even without walking along (kept the phone 10 cm from the BT-dongle) it
kept bouncing: found-gone-found-gone Unworkable.

I wanted not only presence, but speech-patch also via BT.
Idea was, that if co-workers are located at the other end of the world,
they can still be reached on they handy, even when no GSM-roaming is
acceptable (due to costs).

hw


--
_
-- 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] IAX2 support of video

2013-01-09 Thread Hans Witvliet
On Tue, 2013-01-08 at 08:21 -0600, Danny Nicholas wrote:
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry
 Geis
 Sent: Monday, January 07, 2013 6:53 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] IAX2 support of video
 
  
 
  
  
 According to this:
 https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
 yes.
  
  
 
  
 
 
 I have a local server with two video phones - running SIP to each
 phone. Works.
 Then I have an IAX2 connection from that local machine to another
 machine.
 then a SIP connection from that machine to another machine where the
 same model
 video phone is in use. A call to that phone does not show video only
 audio.
 
 All machines have in sip.conf:videosupport=yes
 
 Is there something else to get SIP/IAX2/SIP video call to work?
 
 Thanks
 
 Jerry
 
  
 
 Make sure you have the H.26X codec enabled at all points.
 
 
Video is hard, but to make life easier, it is handy to add an extension
that does the echo-function (after an optional announcement)
That takes video-codec-mismatch out of the equation, as you are talking
to yourself.

Other benefit is that there is always someone that will answer the
phone ;-) And won't complain doing video-cakk when dialing at 3am.

hans



--
_
-- 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] Doubt regarding jabber

2012-12-19 Thread Hans Witvliet
On Tue, 2012-12-18 at 10:01 -0600, Jonathan Rose wrote:
 Harish Mandowara wrote: 
  I have Asterisk server 1.8.19 with jabber enabled.
  
  On the other side i have openfire server with asterisk-im enabled.
  
  I have a doubt, whether my sip client connected with asterisk can
  send message to other sip client, which is connected to same
  asterisk server.
  
  
  I have jitsi as a sip client.
  
  If its possible. Than please suggest any documentation regarding
  this.
  
  any help??
  
  THanks a lot
 
 As far as I'm aware, SIP clients are generally incapable of using
 XMPP to send and receive messages. I'm aware Jitsi can act as an
 XMPP client, but its functionality as one has basically nothing to
 do with Asterisk. Asterisk can use XMPP send and receive messages
 to/from an XMPP server (also clients on that server by relay).
 Jabber is also used for Google Talk and Google Voice, but I'm not
 sure which versions those features work best in. I'd imagine 11
 would be your best bet if you wanted that functionality since it
 has a bunch of Jabber improvements as well as chan_motif.
 
 So if you want Asterisk to send jitsi an IM, you need to set up
 account on an XMPP server for them to use (as well as profiles to
 connect with). Once you are sure you have Asterisk and the Jitsi
 client connected to the XMPP server, you can send the message with
 the dialplan application 'JabberSend' which takes arguments of
 account (which is the account you are using to send), jid (who
 is receiving the message) and the message itself. You can similarly
 receive messages on Asterisk by using the JABBER_RECEIVE function
 with similar arguments except no message and an optional timeout.
 

Hi,
probably Jonathan is correct.
firstly, much changes/improvements has been made since 1.8.19 so you
might give asterisk-11 a try, in which xmpp has got a lot of attention.

Secondly, as you use Jitsi, why don't you use jitsi's xmpp capabilities?
afaicr, the SIP-part of jitsi is only capable of simple, not xmpp.

Hans.

--
_
-- 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] sip-user status

2012-12-13 Thread Hans Witvliet
Hi all,

I'm caught up in a struggle between people how can not make up their
mind... Half way implementing a asterisk farm and they come up with
another feature they've seen in kamaillo.

What he showed me was this: three registered sip users,
a) one changes his presence status on his softphone, and all see the
status change.
b) one calls another, and the third person see the status of the other
two change to busy.

I've seen code/dialplan snippets where you could change your status by
dialling a specific extension, on which asterisk will react (and change
some variables accordingly), but that is not what i'm looking for.

It seems that kamaillo has build-in features to react on sip-simple
changes.
Can i perform the same trick with asterisk? if so, how?


Hans.

--
_
-- 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] [asterisk] Guide for setup a server for end2end video call

2012-12-11 Thread Hans Witvliet
On Tue, 2012-12-11 at 23:02 +0800, Barco You wrote:
 Dear List,
 
 
 Where can I find a guide for setup an Asterisk server which can
 eastanblish a simple video call from two sip clients?
 
 
 Thank you!
 
 
 Regards,
 Barco

Hi Barco,

I don't think there is a specific guide for this.
From the top of my head..
In /etc/asterisk/sip.conf, you will find the the default setting is
_not_ to have video enbled.
So either you enable it gloabally (for all sip-users) or individually
for specific sip-users.

Greatest pit-fall. you have to analyse the codecs for all hard-phones
and/or softphones. You should have atleast one common codec enabled.
If not, you will only get an audio connection (unfortunately, without
any warning)

Most safe option (atleast to begin with) is to use same clients at both
sides, and configure them identically.

Second suggestion, is to define a echo-function. If you dial the defined
extension, you get not only audio echo, but also video-echo.

I've been testing with a couple of clients, with various results.

hans


--
_
-- 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] disabling regular expressions

2012-11-15 Thread Hans Witvliet
On Thu, 2012-11-15 at 12:13 +0100, Frederic Van Espen wrote:
 On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
  In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
  But afaicr the dots will cause problems 
 
 If your extension does not start with an underscore, it is not
 considered as an extension pattern. Correct me if I'm wrong please!
 
yes, indeed. Just tried it: OK.

Stupid of me [blush]

--
_
-- 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] disabling regular expressions

2012-11-14 Thread Hans Witvliet
Hi all,

Is there a simple way of disabling regular expressions in the dialplan?

Reason for asking, is that people hate to remember numbers.
So i want to use there full smtp address as as their extension.

In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems

Hans




--
_
-- 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] Managing complex setups with Asterisk

2012-11-08 Thread Hans Witvliet
On Thu, 2012-11-08 at 10:07 +0100, martin f krafft wrote:
 also sprach Jeff LaCoursiere j...@sunfone.com [2012.11.07.2049 +0100]:
  Just to chime in, if you REALLY want multi-tenant, it is super
  easy and surprisingly efficient to use kernel level virtualization
  to run multiple instances of asterisk (and even FreePBX).  We use
  LXC to do this.  The host runs an instance that has the dahdi
  hardware, drivers, and upstream connections.  The clients have
  SIP connections to the host for all inbound/outbound
 
 Yes, separation into logical units is one way forward, but then you
 will necessarily have redundant configuration between the instances.
 It's nice to have clear separations (unless you cannot clearly
 separate), but I am not convinced that this decreases complexity.

Actually, i would suggest breaking it up and store most of your data
into mysql (realtime).

By breaking up, you can separate distinctive parts, like pstn-gateways,
GSM-gateways, internal-proxies, external-proxies, voice-mail,
conference-server, etc etc.
If you store the user-specific data into a database, it doesn't matter
on which proxy you register, the configuration is shared them all.
Same for your dialplan.

If you use LXC, the overhead will be less compared with using XEN. 
And if you keep each asterisk-container stupid, it is easier to
maintain/replace.

hw


--
_
-- 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] asterisk as IVR using 3g usb modem

2012-10-19 Thread Hans Witvliet
On Thu, 2012-10-18 at 15:45 +, Mahendra Dobariya wrote:
 hi, 
 I want to use asterisk as IVR system ,
 but to make and receive GSM call, i want to use 3g usb modem.(voice
 enabled)
 http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php
 
 
 and i want to install this system on two different machine
 1 on mac os x -
 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/
 
 
 thanx in advance..

Are you very sure about the last one (i.e. the r-pi)?
These have a very few resourses (cpu, mem)
If looking for something small, how about latest pandaboard, a bit more
expensive, but less limited:

http://www.hardware-modules.com/index.php?page=Browseproduct_type=SBCdesigner=Texas%20Instrumentmodule=Pandaboard%20ES%20(Texas%20Instrument%20-%20OMAP4460)lang=en

And still cheaper than most intel-based sff-boards.

hw


--
_
-- 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] asterisk as IVR using 3g usb modem

2012-10-18 Thread Hans Witvliet
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote:
 
 On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
  U would have to write a dahdi module for this 3G modem to help
  asterisk understand it as standard gsm channel.
  
 Look up chan_datacard (i think that's what it's called from memory).
 
 
 Steve
 
It got renamed, and is now:
http://code.google.com/p/asterisk-chan-dongle/

It's just a tgz, no docu, no wiki..
chan_dongle-1.1.r14.tgz  
chan_dongle version 1.1 revision 14 sources
Featured 6 days ago  6 days ago  184 KB  159


Looks a bit fresh...


--
_
-- 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] RC2, was: Motif/XMPP for Google Voice

2012-10-17 Thread Hans Witvliet
Hi,

With regards to:
On Mon, 2012-10-15 at 09:09 -0500, Joshua Colp wrote:
 asterisk asterisk wrote:
  Dear all,
 
 Hola,
 
  I wish to ask a question of the new Motif Channel in asterisk 11.
 
  I successfully compile the binary and run without error. However, when
  dialing out, no external connection  only ringing.
 
 During testing some issues were uncovered with the Motif channel driver, 
 but unfortunately they did not make the last release candidate. My 
 suggestion is to get Asterisk 11 from SVN or if you are not comfortable 
 with that wait until the official Asterisk 11 release.
 
 Cheers,
 

And to: Asterisk 11.0.0-rc2 Now Available

skimming through
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2

I did not see any reference towards Motif/XMPP.
So your code is still only in SVN, not in the RC2?


Hans


--
_
-- 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] conversion?

2012-10-11 Thread Hans Witvliet
On Wed, 2012-10-10 at 18:09 -0300, Joshua Colp wrote:

[snip]

 Yes, there is no capability for video transcoding in any version of 
 Asterisk.

Thanks for pointing out!

So in case my managers starts nagging about it, they have two options:
A) use hard/soft-clients with comparable codecs,
B) raise enough funds for implementing video-stream transcoding ;-)
  [tough job, but do-able]

hw


--
_
-- 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] conversion?

2012-10-10 Thread Hans Witvliet
Hi,

Perhaps can someone tell me if i had the wrong expectancies

If one sip-clinet only supports GSM-codec, and another only supports
g711-U, they still can call each other and asterisk does the transcoding
Correct?

If i try to do the same with an AV-call, (one only h264, the other only
h263) there is no video-transcoding, and you get an audio-call only.

Sure, video transcoding is probably horrible CPU-expensive, but is what
i see correct (on a 1.8 system)?



Hans

--
_
-- 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] motif load

2012-10-10 Thread Hans Witvliet
Hi,

Are there any thoughts about how cpu-expensive motif is?

Does it only translate SIP -- jingle (during call-setup) 
if so, impact will probably neglectible.

or does asterisk remains constantly in between the data-stream?
In that case, it might be something to pay serious attention to, when
doing multiple call conversions simultaneously...

hw



--
_
-- 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] Case-sensitivity of Dialplan variables.

2012-10-03 Thread Hans Witvliet
On Tue, 2012-10-02 at 17:11 -0700, Ira wrote:
 At 02:19 PM 10/1/2012, you wrote:
 So respond here and let me know what you think. I got a couple of replies on
 the -dev list and they said that this would be good to put out on the -users
 list too.
 
 Mark Michelson
 
 In true Republican fashion, I'm going to vote for case-insensitivity.
 
 Given that many of the users were not programmers and didn't likely 
 grow up in a case sensitive world I'd also vote for case 
 insensitivity. I fall into that category, I grew up with dBase, 
 Clipper and VB and case issues get me all the time when I program in C.

I would vote for case-sensitivity.

True, i grew up in the early day's of PDP11, flex, uniflex and so-on,
where case-sensitivity was default.

I think it is a bad habit to write something else, from what you expect.

More important is, that you get a un-avoidable error, when you try to
read a variable, that isn't initialised (due to mixed case).
Like in the old fortran/pascal/C days, where you just get a compilation
error, that you had to solve before you could continue

There is already too much insensitivity in this world,
let's get rid of (at least) case insensitivity!

hans


--
_
-- 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] RealTime table fields ordering

2012-09-28 Thread Hans Witvliet
On Fri, 2012-09-28 at 01:33 -0700, Vieri wrote:
 Hi,
 
 According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip:

[snip]

 So it seems that the contrib directory and the asterisk.org wiki are 
 inconsistent and incomplete.
 Of course I understand that these are 'contributed' files but they should be 
 proof-read by the Digium devs before packing them up into the official source 
 tarball. Or am I wrong about my observations concerning field order and field 
 omissions?
 
 Thanks,
 
 Vieri
 

how about the line:
 `ipaddr` varchar(15) DEFAULT NULL,

Wonder how they try to squeeze an IPv6 address in it...
should be:
 `ipaddr` varchar(50) DEFAULT NULL,


hans

--
_
-- 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] chan_motif, xmpp, jabber, jingle

2012-09-20 Thread Hans Witvliet
Hi all,

For one of my inverstigations it looks like i'm back to square one

I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail. 
No google is involved as i use a local xmpp server (ejabberd)

I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but
some suggested me to have a look at asterisk11,so i did...


I downloaded and built 11-beta1.
Edited (according to the asterisk11 wiki-page) extensions.conf,
chan_motif.conf, jingle.conf and restarted.

Same behavior, except for minor details.
As soon as I start, ejabberd tells me that the defined user becomes
online.
From jitsi I can send a text-message, which I see as I enabled debug
in motif.conf
(This is actually progress, as in 1.8.15.1 I saw only empty strings
coming along ;-)

But when starting an audio or an AV-call, I only see the xmpp-debug
message (used to be jabber-debug-message).
Within de xmpp-message I see the capabilities (samplerate, codecs,
address, port) from the jitsi-client.

Although I made a separate context in the dialplan, it seems never to
get there: hence no answer :(
Eventhough I explicitly point to them in xmpp.conf and jingle.conf
So my client remains in connecting for ever




Other suggestion were to look at the xmpp server.
However, i got three client-machines (two Ubuntu-12.04, one XP) all are
installed with Jitsi (xmpp-client) and multiple accounts that are
registerd on ejabberd. All of them can place jingle-calls to each other:
so nothing wrong with the xmpp-clients or the xmpp-server.

The fact that i see on the cli xmpp-debug messages ariving when trying
to connect, or sending text messages indicates that asterisk is seen by
xmpp-client/server.

But i seem to be be missing something some vital config, as asterisk
does not respond at all at those incoming trafic.


Hans

--
_
-- 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] chan_mobile

2012-09-18 Thread Hans Witvliet
Hi all,

In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)

After i filled in the mac-addresses of the BT-adapter and the one from
my phone, i see it is recognized, got connected, and immediate gets
disconnected.
Same behaviour if i use a completely different phone (BB).
BT on either phone (one at the time) is constantly on and visable'.
Distance between dongle and phne just some centimeters


Any suggestions?

Hans

--
_
-- 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] chan_mobile

2012-09-18 Thread Hans Witvliet
On Tue, 2012-09-18 at 17:43 +0100, Sebastian Arcus wrote:
 Hi Hans,
 

 
 The following page has some useful info:
 
 http://www.voip-info.org/wiki/view/chan_mobile
 
 Sebastian

Indeed. Didn't realise it was so picky.
just bought a couple of bt-adapters.
Will try again tomorrow and feed the results into the wiki..


Tnx.


--
_
-- 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] multiple users for jabber.conf

2012-09-12 Thread Hans Witvliet
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote:
 Hans,
 
 I did not try 10 or 11 as I run 1.8.15.  Following are the related
 conf files.
 
 gtalk.conf
 
 [General]
 context = default
 allowguest = yes ; Required if you want to accept calls
 from people Not on your contact list.
 bindaddr=private IP   ;; These two settings are very critical for
 getting
 externip=public IP  ;; gtalk audio with Asterisk server behind
 NAT
 disallow=all
 allow=ulaw
 
 [guest]   ;;special account for options on
 guest account
 disallow=all
 allow=ulaw
 context=from-trunk
 connection=Jab01
 
 jabber.conf
 
 [general]
 debug=no ;;Turn on debugging by default.
 autoprune=no   ;;Auto remove users from buddy list.
 autoregister=yes  ;;Auto register users from buddy list.
 
 [Jab01]  ;; Label
 type = client   ;; Client or Component connection
 serverhost = talk.google.com
   ;; Route to server
 username = google-user-nam...@gmail.com/asterisk;; Username with
 optional resource.
 secret = Google password
;; Password
 priority = 1   ;; Resource priority
 port = 5222 ;; Port to use, defaults to 5222
 usetls = yes;; TLS is required by talk.google.com,
 you'll get a 'socket read error' without
 usesasl = yes   ;; Use sasl or not
 timeout=100 ;; Timeout on the message stack
 status=available;; One of: chat, available, away,
 xaway, or dnd
 statusmessage = Connected via Asterisk ;; Custom status message
 
 [Jab02]
 type = client
 serverhost = talk.google.com
 username = google-user-nam...@gmail.com/asterisk
 secret = Google password 2
 priority = 1   ;; Resource priority
 port = 5222 ;;Port to use, defaults to 5222
 usetls = yes ;;TLS is required by talk.google.com,
 you'll get a 'socket read error' without
 usesasl = yes  ;;Use sasl or not
 buddy=Buddy1@gmail.com  ;;Manual addition of buddy to list.
 buddy=Buddy2@gmail.com  ;;Manual addition of buddy to list.
 timeout=100
 status=available
 statusmessage = Connected via Asterisk
 
 [Jab03]
 
 [Jab04]
 
 and so on.
 
 Reagrds,
 Vladimir
 

Thanks Vladimir,

Will digg up an 1.8 machine and give it a try!
afaics the only diference is that i am using a local xmpp server
(ejabberd) instead of google, but that should only make things easier i
think...

Hans


--
_
-- 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] multiple users for jabber.conf

2012-09-11 Thread Hans Witvliet
Hi all,

Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and
11 version of asterisk.
In each example i got the impression that the asterisk server is
registering on a XMPP server as a single user with the credentials as
specified in jabber.conf.

Instead of a single xmpp-user, could that also be multiple users?
For instance, for each sip-user an xmpp-user?

When i skim through most of the examples, the asteriskbox is used for
making an outbound call with the jingle protocol.


But how about incoming calls?
I presume you need multiple xmpp-accounts, in order to differentiate
multiple destinations. Not?

Or to describe it in an other way: If you just do a single
xmpp-registration, how can you become a destination for different
end-users? how about multiple presence-states?


[utterly confused] Hans

--
_
-- 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] asterisk on arm

2012-09-04 Thread Hans Witvliet
On Tue, 2012-09-04 at 13:58 +0500, qasimak...@gmail.com wrote:
 How about stripping it down to bare minimum's?
 

How about an other ARM-board?
http://gooseberry.atspace.co.uk/?page_id=13

Specifically the more mem (4GB) will help..

hw



--
_
-- 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] xmpp / sip

2012-08-24 Thread Hans Witvliet
Hi all,

After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(

They want to have an Ejabberd server, with xmpp-clients.
When you see a contact coming online, just point and click for making a
phone call.

Sounds/looks nice and do-able, but there is one catch:
-incoming / outgoing call towards corporate pstn (E1)
-incoming / outgoing call towards public pstn (E1)
-incoming / outgoing call towards GSM (nanobts)

So does anybody any any thoughts about mixing/translating XMPP and SIP?
firstly: sip - xmpp
If i get an incoming call for user-A, i should be able to see the status
of user-A, and if available, pass the call on to him.

SecondlyL xmpp - sip
So if i can define a external-user-A and let him login on ejabberd, he
should indicate xmpp-available, and local people should be able to
call him in that way.

Sounds rather complicated, i know...


Hans


--
_
-- 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] Asterisk hangs while starting in OpenSuse 12.2

2012-08-23 Thread Hans Witvliet
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote:
 Hi,
 
 
 Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1
 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed,
 installation went fine. 
 
 

Have you tried the versions from the OBS?

Or perhaps a virtualbox issue? Its notorious for vapourizing
cpu-cycles...

hw



--
_
-- 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] HP DL360 G5 better than HP DL360 G7 ?

2012-08-01 Thread Hans Witvliet
On Wed, 2012-08-01 at 19:39 +0800, D Tucny wrote:

snip

 
 For reference... In my opinion HP servers should never be bought
 without the battery or alternative, they shouldn't even be offered for
 sale without it...

In my case, our purchase department changed our order.
They thought in their infinite wisdom, that they could strike a better
deal (without informing us of course).

During the two years to find who to blame, the all the G7 were replaced
by a couple DELl's with fusion-IO cards, 64 cores and 256GB mem.

Don't think we go back to the G7's anymore ;-)

hw


--
_
-- 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] HP DL360 G5 better than HP DL360 G7 ?

2012-07-24 Thread Hans Witvliet
On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote:
 Any tips on solving the following performance conundrum:
 
 Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
 
 tcpdump running to capture UDP 5060/SIP signaling to .pcap files
 
 All calls are ultimately B2BUA client - asterisk - PSTN
 
 Media stays on Asterisk at all times
 
 AGI script has exit handler that connects and updates an external
 database upon BYE from either side.
 
 I know that if exit handler hangs around too long, Bad Things (tm) will 
 happen.
 
 Oddly, under load (60-100 B2BUA calls), the G7s start complaining:
 
 Autodestruct on dialog 'CALLID' with owner in place (Method: BYE)
 
 I/O wait is actually higher on the G5s, the G7s have fancy disk cache
 cards and never get above 1% i/o wait
 
 turn off the tcpdump process on the G7s and Autodestruct warnings go
 away.  The G7s should have
 much more capacity than the G5s but we never, ever get Autodestruct
 ... Method: BYE on the G5s.
 
 OS is identical CentOS in both cases.   Every other environmental
 config is the same (network, subnet, DNS etc).
 
 Architecture/bus/network card difference?  tcpdump starving some other
 resource to cause stuff to slow down?

Hi Tom,

Regarding G5's and G7's
We have them both (all though not for asterisk) at work. and found a
little snag.
The purchase department interferred with our initial ordering of the
hardware. As a concequence, they did not order the battery for the raid
controller. (seems you have to order that explicitly)
After it arrived, we were able to install linux on it and use it, but
disk-io was way slower that the original G5.

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

Hans



--
_
-- 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] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk

2012-07-24 Thread Hans Witvliet
On Tue, 2012-07-24 at 11:07 +0530, Kannan wrote:
 Hi Stelios,
 
 
 Thanks for the response. 
 
 
 I take the following excerpt from your response. --- You can, but
 usually for virtual/hosted pbx's you need an additional
 layer of management software or a lot of copy paste
 
 
 Could you please elaborate on that? Do need to modify Asterisk or
 there exists some software that does the job?
 
 
 Regards.

If you have a (large) number of asterisk-servers, it might be handier not to 
duplicate configs,(to avoid mismatches) 
but have sip and exension config centralised in ldap or mysql.

And when using dns-round-robin, you can almost add/remove machines on
the fly.

hw


--
_
-- 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] asterisk 1.8 on Solaris/sparc

2012-07-19 Thread Hans Witvliet
On Wed, 2012-07-18 at 02:27 -0400, Jeremy Kister wrote:
 I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
 
 The system itself is happy and phone calls (between two parties) seem fine.
 
 Unfortunately, when a caller listens to a Playback recording, there 
 seems to be moments of stutter - perhaps 1 second of stutter for every 
 10 seconds of Playback.  The stutter is not consistent at the same point 
 of the playback file.
 
 To eliminate encoding as an issue, I have only codec_ulaw/format_pcm 
 loaded and the recording is ulaw.  I've niced down the asterisk process 
 to -19 even though I don't see asterisk taking more than 3% cpu.
 
 
 Is this behavior indicative of a timing problem?  loading 
 res_timing_pthread.so makes things horribly worse.  i don't believe any 
 other software timer is available for Solaris/sparc, right ?
 
 other thoughts ?

Perhaps system too busy, disk not fast enough?
before doing a play-back, run iostat 1 in another window

Incase iowait is too high, try moving the files with the playback
sound/speech upon tmpfs (thus eliminating the hard disk)

hw


--
_
-- 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] AMR - Segmentation Fault

2012-07-05 Thread Hans Witvliet
On Wed, 2012-07-04 at 10:15 +0530, Chandrakant Solanki wrote:
 So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch
 also works in 1.8.13.0??
 
 On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl
 wrote:
 On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
  Hi All,
 
  OS : Cent OS 5 64Bit
  Asterisk : 1.8.0-rc2
 
  AMR Source Link :
 http://sourceforge.net/projects/aterisk-amr/files/
 
  When I tried to call or start asterisk, I found
 Segmentation Fault.
 
 
 Without trying to be pedantic, but 1.8.0-rc2
 Ever considered upgrading? To 1.8.13.0 or so..
 

Hm, i see.
Looks like somebody is seriously hibernating:
1.8.0-rc2_asterisk_amr_patch.diff 2010-10-15 Almost two years old!

If it is only the codec itself, you might try:
http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Multimedia/src/amrnb-10.0.0.0-1.1.src.rpm
 
or
http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Essentials/src/amrwb-10.0.0.0-1.1.src.rpm
 

As these are source packages, you might be able to turn then into deb's
or so


Hans

--
_
-- 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] AMR - Segmentation Fault

2012-07-03 Thread Hans Witvliet
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
 Hi All,
 
 OS : Cent OS 5 64Bit
 Asterisk : 1.8.0-rc2
 
 AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
 
 When I tried to call or start asterisk, I found Segmentation Fault. 

Without trying to be pedantic, but 1.8.0-rc2
Ever considered upgrading? To 1.8.13.0 or so..

hans


--
_
-- 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] port 5060 is blocked by ISP

2012-07-01 Thread Hans Witvliet
On Sun, 2012-07-01 at 13:04 +0530, alok srivastava wrote:
 dear
 i have configured properly asterisk. At the one end i am using x-lite
 soft ph and another end twinkle. call is going properly from both end
 but after picking the phone not able to listen other one.
 when i checked the port 5060 on the asterisk server it is always
 showing closed while i have flushed all the rules from iptables
 (iptables -F)
 
 PORT STATE  SERVICE VERSION
 5060/tcp closed sip
 
  telnet localhost 5060 (could not connect)
 
 regards
 alok

Hi Alok,

telnet is a very crude tool to test with.
Try hping or nmap instead.

Hans


--
_
-- 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] fritzbox

2012-06-18 Thread Hans Witvliet
Hi,

Couple of moments ago my asteriskbox with a bri-card went down.
(burn-out)

I've heard that it seems to be possible to use an fritz!box as an
isdn-gateway (isdn -- sip)

Anyone around who has good/bad experiences with those AVM-boxes?

(yeah, i know it is tech overkill, but i'll get an dualband wifi router,
that is Ipv6-ready with it)

Hans.


--
_
-- 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] looking for solid state like PC suitable for Asterisk

2012-05-10 Thread Hans Witvliet
On Thu, 2012-05-10 at 07:40 -0500, Tim Nelson wrote:
 - Original Message -
  On Thursday 10 May 2012, Bart Coninckx wrote:
   I'm looking for a smaller,
   appliance-type like PC, preferably solid state and fanless PC.
   Since it's only going to run Asterisk for a couple of extensions I
   don't
   think CPU and RAM need to be maxed out.
  
   Does anyone have inspiration/experience for/about such a model?
  
  Raspberry Pi would be the obvious choice, surely?
  
 
 The hype around the Raspberry Pi is enormous. I would not consider it a real 
 option for production voice until it's had a chance to mature and be 
 available for some time to iron out the bugs, both hardware and software 
 related.
 
 My $0.02 USD.
 
Another couple of cents:
the pi comes only with arm-cpu and limited amount mem - no upgrade
possible.

Might be an issue for asterisk...

have a look at: http://www.fit-pc.info/
As long as you don't need to plug in a pci-board it is nice small and
uses hardly any amps.

hw


--
_
-- 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] using Wifi smartphones as SIP clients

2012-05-07 Thread Hans Witvliet
On Mon, 2012-05-07 at 19:03 +0100, Roger Burton West wrote:
 On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote:
 What about phones like the Unidata WPU-7800 (
 http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
 experience with those? Would these also suffer from connection
 losses?
 
 I've been using a UTStarcom GF-210 for the last year and more as my
 personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP
 is slightly better than 2G, getting it to talk to Asterisk is no problem
 at all, but certainly if you're moving from one wifi device to another
 you will get dropped calls. If that's your use case, it's going to be
 that way whatever hardware you use - I haven't seen any implementations
 of 802.11F or 802.11r in the field.
 

Hope that these are better that the utstar F1000:
Keep on re-chargibg as battery is empty in no-time, and security is
lousy; just  wep, no wpa.

hw


--
_
-- 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] Commercial SSL certs on Asterisk 1.8.10.0 with Polycom phones for encrypted calls using TLS and SRTP?

2012-03-08 Thread Hans Witvliet
On Thu, 2012-03-08 at 16:50 +, Gavin Henry wrote:
 
  Ah, this makes sense now. So as of today the status of TLS and SRTP in
  anything
  other than 1.4.X is unknown?
 
 
  Umm... no :-)
 
 OK, sorry :-)
 
  Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
  these were tested with Polycom phones the last time we did interop testing
  with those phones.
 
 Ah, I forgot when it was added.
 
afaicr, it was in 1.6.2
  The status of SIP/TLS and SRTP support in the Asterisk releases that have
  them are not 'unknown'; they are there and expected to be working. I was
  just pointing out that Digium has not specifically tested Polycom phones for
  interop with these features, and certainly has not specifically tested usage
  of TLS certificates issued by any particular CA.
 

btw, commercial certs are not so special.
Somewhere in the chain (root-ca), there is a self-signed cert.
You can make such chain yourself, 
root-ca - sub-ca - sub-ca and finally a server+client cert.
Or, you can get a free cert from cacert.org

hw


--
_
-- 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] Proposed changes to Asterisk release and support cycles

2012-02-01 Thread Hans Witvliet
On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote:
  I like the idea of LTR release more often that would have the
  feature patches baked in.  Case in point the new conference app
  requires a jump to version 10 while the 1.8 conference app is quite
  useless but 1.8 is my LTR version so I am stuck without the
  conference app in my mainline systems for two years. 
 
 Well said!  This is also true of any type of long term supported
 release whether if it's an operating system, application, etc.  In the
 LTS name, it conjurs up thoughts of Ubuntu, but comparisons to
 RHEL/Fedora are far more appropriate I would think as Ubuntu focuses
 nearly exclusively on new point releases while backporting new
 features is what a company like Red Hat excels at and should be the
 prime example of how to run dual software channels (enterprise release
 in RHEL vs. hobby release in Fedora). 
 


 
 I know distros and applications are two fundamentally different
 things, with entirely different goals and requirements, but I still
 think Red Hat provides the best example because 1) they have turned it
 into a science how smooth their development process goes in ratio to
 satisfied customers and 2) it's the only other open source software
 project I can think of that can accurately compare.  In a past meeting
 I had with Digium while working for another company, they too directly
 drew a correlation between the new LTS idea and ubuntu lts/non-lts and
 rhel/fedora.
 
 The conference app changes since 1.4 I haven't been thrilled with, but
 in the whole time I've been supporting 1.8.x for my customers, I've
 come up with a very stable solution building on it and I haven't had
 any surprises come my way.   
 
very well said indeed.
Some (...) distro's think dat LTS implies a complete feature freeze.
Others are more flexibel about it, that besides current versions of
applications, they are willing to support both elder _and_ newer
versions. (as example, i'm refering to the fact that hours after the
anouncement, firefox10 became available for sles11)

As said, re-written features like conference, are that important that
one shouldn't have to wait years for the next LTS. So this overlap of
multiple LTS-versions looks very much attractive

Having said that, i do understand that multiple versions of
features/applications puts an huge extra burden on the people who have
to maintain both versions, as the original version (as the term LTS
implies) should be maintained with all its limitations also.

hw

--
_
-- 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] best softphone for 2012?

2012-01-07 Thread Hans Witvliet
On Fri, 2012-01-06 at 16:00 -0600, Tom Poe wrote:
 Just installed asterisknow 1.6.  I can access freepbx.  I need to test 
 system on my LAN.  Which softphone is best to use?  I'm running ubuntu 
 on Dell optiplex G260 desktop at home.  I'm hoping to setup basic IP PBX 
 for incoming/outgoing calls.  No video.
 Tom
 
Hi,

Our requirements were different, so we came to three candidates;
linphone, ekiga and jitsi
Linhphone is easy to pre-configure from a script and the buttons are
easier to use, but lacks the possility for an ldap-adres-book.

With ekiga you have the adresbook, but you have to use the mouse
everywhere (the return-button gives unexpected results)

And with jitsi (java-based) you are independant of Qt/GTK.

hw

--
_
-- 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] Simple Generic IVR to get us up an running Quick

2011-12-06 Thread Hans Witvliet
On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote:
 Hello Everyone,
 
 Are there any descent generic IVR recordings, that we can
 use to quickly get our PBX up and running? It will obviously
 not include the company name.

It's easy enough to make your own recordings.
Word of caution though.

It might be advisable to ask somebody outside the company to record the
phrases, Wonder why?

At home i did it my self, and i still hear people stating that they have
been talking at me, totaly unaware that it was just the voicemail
anouncements. Peope just hear a voice, but seldom listen.

And not just 90-old aunts, 
But people from helpdesks and even CEO's.

Sometimes i wonder, if i should ask/test the callers I.Q. ,
And adapt the IVR's accordingly  ;=)

hw

--
_
-- 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] A new hack?

2011-12-06 Thread Hans Witvliet
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
snip

 Your security needs depends on your environment. At this point in time, 
 all of the hosts I manage for my clients exist in very limited 
 environments and have very small attack surfaces. They are racked in 
 secure data centers. They only accept SIP from clients with static IP 
 addresses that we have an existing business relationship with. They only 
 accept SSH connections from me. They only accept HTTP connections from me 
 and my boss. That's about it. I don't see where F2B adds much value for 
 me.
 
 *) Lots of admins think they can't limit access to servers because they 
 have 'mobile' users. Your users probably don't need to access your servers 
 from every single place on the Internet. If your users don't come from 
 China, North Korea, Iran, etc, you can block entire regions with a few 
 rules and eliminate 80% of probes and attacks from reaching your servers 
 in the first place. Apologies in advance if you happen to live in some of 
 these regions -- feel free to `s/China, North Korea, Iran/United States, 
 Canada, England/g`
 

Perhaps an other suggestion.
If they are true road warriors, i presume they are capable of setting
up an vpn to the company.
In that case, only allow  registrations/calls through the secured
tunnel. Then it's not any concern to asterisk.

And if they can breach your tunnel, you have something else to worry
about.


hw

--
_
-- 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] how to find out one way latency

2011-12-01 Thread Hans Witvliet
On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote:

 
 You can make a pretty good prediction with ping.
 sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation 
 of voip traffic.  let it run for awhile, then press ctrl+c and see how 
 many packets were dropped and also check the mdev number.  If mdev is 
 low and packet loss is almost nothing then you can expect decent voice 
 quality.  It may not be a 100% perfect test, but I'll bet you a vast 
 majority of the time I can do that test and tell you whether it's going 
 to suck.
 
 latency by itself with low jitter and no packet loss just means delay.  
 It's a matter of opinion and circumstance how tolerable delay is, but I 
 think your 230ms ping is at the upper edge of what most people can live 
 with.  Much more than that and you'll be tempted to say 'over' at the 
 end of sentence.
 
 --
Fully agree,

Actually, you can do better than just a ping, but it takes some time,
equipment and experience:

What you can do, is adding an extra box inbetween your voip-client and
voip-server, and introduce all kinds of real-life circumstances.
I mean artificial delay, loss, resequencing, duplicating packages,
reduced bandwith. We've done it some time ago as an satelite simulator
You can build it aroud any *bsd/linux box with multiple nics.

The basic idea's you can find at http://lartc.org/
If you combine it with the echo function from asterisk, you can decide
for yourself what it acceptable and what not.

For one of my projects i push the echo destination as the default sip
connection to their soft phone, as i noticed that people at the other
side of town regularly have a worse connection then people using umts or
satelite. Main culprit (in my case) is ill-configured WIFI-setup.
Latencies of over 10,000 ms and loss of 80% are daily events.
And people complaining


hw


--
_
-- 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] Skype For Asterisk (SFA): any replacement?

2011-12-01 Thread Hans Witvliet
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
 On Thursday 01 December 2011, gincantalupo wrote:
  Hi all,
  
  any idea about how to replace Skype For Asterisk?
  
  Thank You.
  
  Giorgio
 
 1.  Migrate your Skype users over to a better product which supports proper 
 open standards. 
perhaps you missed it, but the installed base of skype is unfortunately
slightly (,,,) larger than the amount of peope that are using a decent
product. Alas


 2.  Write to your elected representatives asking that they order Skype to 
 release documentation on their protocols to allow third party 
 interoperability  
 (as is already required under EU law). 

3. make it a offence to use any closed source products like skype. ;-)
Huge fines, jail centences or worse.
[How about an appendice to the Thora, Quran or Bible, even better,
forbid it by the sharia]



--
_
-- 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] how to find out one way latency

2011-11-30 Thread Hans Witvliet
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
 Is there anything else that I should be concerned about, when looking
 to signup for a SIP provider. ?? 
Latency is important, but packet loss also, likewise packet re-ordering.

hw

--
_
-- 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] State of Asterisk+Virtualization+Timing

2011-11-09 Thread Hans Witvliet
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote:
 Is anybody using pci-passthrough?
 
Yes, though quite a while ago.
About three years ago, i used pci-passthrough to give a dom-U access to
a localy mounted smartcard.
But i have a vague feeling that you are up to something else...

I know that forwarding has been done for ethernet and even VGA-cards,
the mere idea of forwarding a analogue or PRI card is quite something
else: Timing is here essential..

hw

--
_
-- 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] State of Asterisk+Virtualization+Timing

2011-11-08 Thread Hans Witvliet
On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote:
 That sucks! What about KVM or XEN?
 
 Nick.

No problems here with XEN.
(Perhaps i should mention, that i use paravirtualsisation to get the
best performance. 
Distro: mix of SLES11sp1 /open_11.4)

hw


--
_
-- 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] State of Asterisk+Virtualization+Timing

2011-11-03 Thread Hans Witvliet
On Tue, 2011-11-01 at 12:08 -0500, Tim Nelson wrote:
 Greetings-
 
 I'm about to dive into the process of virtualizing some of my Asterisk 
 (primarily 1.4.x) infrastructure. In the past, when looking at virt 
 solutions, the primary issue preventing me from moving was the lack of proper 
 timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like 
 to use either OpenVZ or KVM, but each seem to have independent issues that 
 need to be addressed:
 
 OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant 
 access to host node timing source (physical device, or dahdi_dummy in 
 /dev/dahdi/) to the containerized Asterisk process.
 
 KVM - Higher overhead, easier installation, 'true virtualization'. Primary 
 issue is not timing per se, but KVM scheduling. Timing source, while present 
 from dahdi_dummy natively may still not get proper scheduling by KVM process. 
 This could also affect general call quality (even non IAX2 trunked voice), 
 DTMF, etc.
 
 I have to believe there are others running virtualized Asterisk installations 
 with some degree of success on OpenVZ or KVM. Care to share your thoughts?
 
You mist out one more mature virtualization technique: XEN
Virtual machines can use  both hardware- or paravirtualization.
I have used both asterisk (1.4, 1.6.x and now 1.8) to separate machines
where people should do their sip-registration (internet / intranet /
pstn-gateway) and the actual dial-beast.

Main advantage for virtualization is (besides easy scaling) that you can
perform an upgrade in no-time (one VM-machine down, other up) Don't like
it: back in seconds!
Migration with an asterisk on real hardware takes much more resources.
Both in iron and in time.

hw

--
_
-- 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] Asterisk Security: Allow only one phone per sip registration

2011-10-14 Thread Hans Witvliet
On Fri, 2011-10-14 at 10:02 +0300, Muro, Sam wrote:
 Hi there
 
 Consider this. You have three SIP extension 200, 201 and 202 and you have
 configured your phones, say Polycom 331 to those accounts. 200 being one
 very sensitive individual.
 
 Lets say, an insider, get a new phone or perhaps an xlite and configure it
 with the same extension, 200. Asterisk will register it as 200 to the new
 IP address.  Now extension 202 call 200. The hacker answers it and pretend
 is the same person. Do what he want to do and thats it.
 
 Question;
 How can i stop this type of threat
 
 Regads
 Peter
 
Perhaps use secrets?
afaicr the secrets you have to provide for hardphone and softphone are
readonly.
If you avoid something like secret or welcome or the involved
hostname, but instead use a 15 char long generated pwd, he'll have a
long time trying all the possibilities And different pwds for each
phone.

hw

--
_
-- 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] realtime goto/gotoif/dial

2011-09-14 Thread Hans Witvliet
On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote:
 On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote:
  Hi all,
  
  I presume i made a silly mistake while filling a database
  
  But while googling on the results, i came across a lot of messages about
  the layout of app_data in case of goto and dial statements.
  (mostly about using the old | seperator instead of the , separator.
  
  So i was wondering, is this issue been solved? (I presume so, but can
  not find any confirmation about it)
  
  Hans
  
  --
 Hi
 
 It's pipe in 1.4 and comma in 1.6 and 1.8
 

Thanks for confirming it.

As i'm using 1.8.5 i'm now certain that it is a case of PEBKAC
The symtoms are the same, but obviously a different cause. ;-)

Hans

--
_
-- 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] realtime goto/gotoif/dial

2011-09-13 Thread Hans Witvliet
Hi all,

I presume i made a silly mistake while filling a database

But while googling on the results, i came across a lot of messages about
the layout of app_data in case of goto and dial statements.
(mostly about using the old | seperator instead of the , separator.

So i was wondering, is this issue been solved? (I presume so, but can
not find any confirmation about it)

Hans

--
_
-- 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] Distributed device state / presence info??

2011-09-05 Thread Hans Witvliet
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote:
 On 09/01/2011 04:39 PM, Hans Witvliet wrote:
 
  From the asterisk-bible and the wiki's i learned that it is possible to
  let asterisk do some of the presense-info by means of the jabber.conf
  file and a seperate xmpp-server.
 
 You are misunderstanding a bit; Asterisk can use an XMPP server and 
 PubSub to *distribute* presence information among a cluster of Asterisk 
 servers. This information is not intended to be directly sent to XMPP 
 clients.
 
  What i assume (please correct me if i am wrong) is that when a client
  registers/deregisters, asterisk will update the presence info towards
  the XMPP-server. Correct?
 
 Yes, in order to let other Asterisk servers in the cluster know about it.
 
  But otoh, what people would like to see is who is on line.
  And not only on the asterisk-server that they are connected to, but also
  from other possible asterisk servers.
  And furthermore, each registered user might want to set their
  presencse-status to either free/busy/away/what-ever.
 
 Asterisk does not support 'user' presence; it supports device and 
 extension presence. In some applications these can be used 
 interchangeably, but in others they don't match up very well.
 
Ok, so that should mean that the presence-status is controlled by the
fect wether a sip-user is registered or not?
If so, i'm still making a mistake somewhere:

I tried to simplify my configuration: Just one asterisk machine, 
And two users (my and myself) with two linphones, one from a XP-machine
and the other from a SuSE-machine.

They can both register and call each other, and in linphone presence is
enabled.
However if i manually add on each side the corresponding info (name +
sip-address) in the linphone app, they always remain grey / away Even
while in the middle of an connection to each other.

(Probably not relevant, but i'm using 1.8.3)

hw

--
_
-- 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] Distributed device state / presence info??

2011-09-01 Thread Hans Witvliet
Hi all,

Last couple of days i've arguing with my colleges about presence-info.

From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.

On the other hand, most soft-phones are capable of doing something
with presence, allthough most of them use SIMPLE-protocol, instead of
XMPP.

So if when should one use the presence info from asterisk and when use
the presence info from the softphones.
It looks to me like doing the same job twice.

What i assume (please correct me if i am wrong) is that when a client
registers/deregisters, asterisk will update the presence info towards
the XMPP-server. Correct?

But otoh, what people would like to see is who is on line.
And not only on the asterisk-server that they are connected to, but also
from other possible asterisk servers.
And furthermore, each registered user might want to set their
presencse-status to either free/busy/away/what-ever.

So if the changing/reading is to be done on a softphone, what is the
point of having asterisk doing someting with the device-status???

While writing, i've got a distinct feeling i'm understanding less by the
minute ;(


Anyway, what i'm building is a central server and a number of
asterisk-boxes that act as proxy/six-iax-converter.
All of the registered users should be able to see the presence of all
the users on either proxy.

Hans

--
_
-- 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] Anyone using Asterisk on VirtualBox ?

2011-09-01 Thread Hans Witvliet
On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote:

 
 
 My main interest of being on Virtual platform is portability / Backup.
 In case of any h/w issues, or crashes, simply copy the VM on to
 another box and you are up in minutes.
 
 
 Sanjay
 --
Doing that right now, although in my case i use XEN.
Besides being hw independant, it is easier to play with a different
version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able
to switch back in minutes.

hw

--
_
-- 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] Looking for ideas for nice **Asterisk** home phone system

2011-08-31 Thread Hans Witvliet
On Fri, 2011-08-26 at 19:03 -0400, Eric Wieling wrote:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
 Sent: Friday, August 26, 2011 6:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Looking for ideas for nice **Asterisk** home 
 phone system
 
 On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote:
  On Fri, 26 Aug 2011, linux guy wrote:
  
   How much power does the home asterisk box need ? 
 
 I use a small box (like those hp thin clients) But these are a bit stronger 
 aluminium housing, instead of plastic, and better foor cooling.
 
 Power consumption: 8 Watt under full load
 CPU:  Model: 6.28.2 Intel(R) Atom(TM) CPU Z530   @ 1.60GHz
 Memory Size: 1 GB
 Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet 
 port, others have two
 Size: 10x10 cm
 
 Is this a custom build box or does a company sell them preassembled?We 
 are always on the lookout for potential boxes we can use for small 
 installations.
 
It is pre-assembled,
You can opt for either no internal disk small (8GB) sdd, larger (64GB)
sdd or ordinary disk, And either no, MS, or ubuntu pre-installed.
Also with/without wifi antenna.

hw

--
_
-- 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] phone + video

2011-08-31 Thread Hans Witvliet
Hi all,

I know that a lot of people have negative experiences with
grandstream-2000, but personally. i'd only the repace one poweradapter
after three years...

So, can anybody give some comment on one of their recent models,
the GXV-3175 (the one with the 7 display)
I'm looking for a phone with video capabilities, as i don;t want to
limit my self to testing with softphone..

HtH, Hans

--
_
-- 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] Looking for ideas for nice **Asterisk** home phone system

2011-08-26 Thread Hans Witvliet
On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote:
 On Fri, 26 Aug 2011, linux guy wrote:
 
  How much power does the home asterisk box need ? 

I use a small box (like those hp thin clients)
But these are a bit stronger aluminium housing, instead of plastic,
and better foor cooling.

Power consumption: 8 Watt under full load
CPU:  Model: 6.28.2 Intel(R) Atom(TM) CPU Z530   @ 1.60GHz
Memory Size: 1 GB
Disk /dev/sda: 64.0 GB, 64023257088 bytes
This model has just one ethernet port, others have two
Size: 10x10 cm

hw

--
_
-- 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] Strange network issue

2011-07-22 Thread Hans Witvliet
On Fri, 2011-07-22 at 10:58 -0700, Dave Platt wrote:
  They've got a bunch of Grandstreams that seem to be rock solid... until 
  7:00pm.  At 7:00, some of the phones become unavailable, and stay down.  
  Call 
  quality is solid almost all the time.  But right at 7:00, things go bad.  
  Only 
  some of the phone lines go down and they stay down until the phone is 
  rebooted.
  
  I'm not even sure what to look for when I go to the site.  Any ideas?
 
 I'd look to see if there are any electrical circuits (lights,
 fans, etc.) which are on a timer of some sort, and are automatically
 powered off at 7 PM.
 
 If somebody mistakenly plugged a piece of network kit into such a
 circuit, it would lose power at that time, and your network might
 end up being partitioned, or routing (switch or IP-level) might
 change abruptly.
 

Hi,

Even if there is no equipment you own controlled by a timer, you still
can suffer from it.

Some power companies have different rates for power you use during
daytime or at night.
So even if _you_ don't have equipment on a timer, your neighbours might
have. Something like electrical boilers or so, or other heavy
equipment. Switching them on/off can cause huge spikes on the
electrical wires.

A couple of neigbours at work have their own micro-power-generators.
About one in ten times, when they start delevering power to the grid,
all of our test-systems go down. Only the systems behind the
re-generated UPS (that removes spikes from the powerlines) are protected
against them.

So nasty litte spikes are harder to detect/tracedown than a full
blackout.

hw


--
_
-- 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] dialplan: all extern, except

2011-07-15 Thread Hans Witvliet
Hi all,

Perhaps a no-brainer, but i think i am making my dialplan on my proxy
too complicated. 

Normally, what you find in the examples is that you have to dial a
specific number, other 9 or 0 for an external line.

What i want to do is this:
If you pre-pend a number with something like * then you can dial local
defined numbers, otherwise everything goes through my iax-trunk-line.

So for instance:
*#1 gives you a local welcome text
*#2 gives you the local echo function

while
#1 gives you a remote welcome text
#2 gives you the remote echo function

And ordinary numbers or sip's go straight extern:
0174539053 or j.witvl...@a-domani.nl should go to my main
asterisk-server.


Currently i'm doing it pattern-matching all numbers, and each upper
+lower case character, but i wonder if it can be done simpler.

Hans

--
_
-- 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] realm question

2011-07-05 Thread Hans Witvliet
Hi all,

Trying to find where i got wrong in my config

Is the realm parameter in sip.conf only used for possible
autentication?

The thing is, i got my box more-or-less working as i wanted,
but i can only reach internal functions (like echo-test and so on) and
other sip-clients if i dial 1234@fqdn, while i was expected to be able
to just dial 1234

I presume i have either a mismatch between how the softphones register,
and my asterisk conf.

Kind regards, Hans


--
_
-- 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] realm question: solved

2011-07-05 Thread Hans Witvliet
On Tue, 2011-07-05 at 12:33 +0500, Faisal Hanif wrote:
 The problem you are reporting is not related to realm but can be context or
 domain.
 
Tnx,
It was indeed a domain issue.
In some cases static definitions in /etc/hosts is not a good replacement
for DNS...

hw

--
_
-- 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] Solved, was: Permanent restart after upgrade

2011-06-10 Thread Hans Witvliet
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote:
 On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote:
  On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
  On Thu, 9 Jun 2011, Hans Witvliet wrote:
 
   I went originally from a almost working machine running:
   asterisk180-1.8.3.2-87.1
  
   To a machine that continuously restarts asterisk (+core dumps) running:
   asterisk180-1.8.3-85.2
 
  Any chance you have a mix of Asterisk and module versions? Was
  Zaptel/Dahdi compiled with the proper set of headers for your kernel?
 
  Can you start Asterisk from the command line instead of the usual startup
  script? What do the first couple of errors look like? Capturing the output
  via the 'script' command will help.
 
  For example*,
 
script foo
sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\
-c -d -d -d -f -g -n -p -q -v -v -v
exit
 
  Can you turn off auto module loading and start with no modules?
 
  *) I'm a 1.2 Luddite, so the command line arguments may have changed...
 
  No dahdi/zaptel involved.
  I'll be off to work in a while, report back later.
 
 
  hw
 
 
 It amazes me when people run into a problem but refuse to post logs or
 verbose when you start Asterisk.  Nothing meaninful.
 
 I would wager a gentleman's bet that I can have your system working
 just fine in a half hour or less (unless your bandwidth sucks).
 
 If I do it, then you have to post to the list and you owe me a favor,
 plus, in the future you have to help someone else.
 
 If I don't, I have to post my failure to the list and I owe you a favor.
 
 I have spare cycles, just let me know.
 
 Thanks,
 Steve Totaro

Hi Steve, thanks for your time and consideration.
Hadn't a chance to report back, as i just returned from work ;-(

I think i found the reason behind it; a missing file from the update.
As the machine involved is not connected to Internet, each and every
file has to be put on a portable medium, checked, and only then i'm
allowed to put it on our corporate lan.
It turned out, that not all required files were transferred on the
usb-disk, or removed by someone. Anyway it, i copied the missing file
( 
../repo/network:/telephony:/asterisk/SLE_11_SP1/x86_64/asterisk180-1.8.4.2-90.1.x86_64.rpm
 )
manually, updated it again and: voila

So regarding not posting config/log/trace/core's..
Initially i just put the symptom's on the list.
Next step would have been any specific log's or config files.
As you know, any most of the logfiles for *, can be rather long, same
for logfiles.

If you are still interested in it (educational purposes) i can still get
the /var/log/asterisk/messages file, and put the relevant section on the
list

Bottomline is however, when doing an upgrade, and only some of the
asterisks RPM's are processed, you get funny results
And what you see on the asterisk console/logfile does not indicate
directly what went wrong.

If i run into a situation, it is mostly that i can not get something
working in the first place, or i made an incorrect change. In those
cases the remedy is obvious.
In this case however, the cause is not PEBKAC or asterisk issue, but
something with the prebuild packages, i'll guess a missing dependancy,
which allowed some asterisk files to be updated, while not all were
present, resulting into an unstable result.


So, i'll try to find the person responsible for packaging, and try to
convince that he ahs some work to do.



Kind regards, Hans

--
_
-- 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] Permanent restart after upgrade

2011-06-09 Thread Hans Witvliet
Hi all,

I got three asterisk-machines, two of them acting as proxies.
On one machine (sles11sp1) i got iritating messages about not bing able
to find codec's and other stuff, so i thought it might be time for an
update: Stupid!

I went originally from a almost working machine running:
asterisk180-alsa-1.8.3.2-87.1
asterisk180-dahdi-1.8.3.2-87.1
asterisk180-1.8.3.2-87.1
asterisk180-odbc-1.8.3.2-87.1

To a machine that continuously restarts asterisk (+core dumps) running:
asterisk180-alsa-1.8.4.2-90.1
asterisk180-dahdi-1.8.4.2-90.1
asterisk180-1.8.3-85.2
asterisk180-odbc-1.8.4.2-90.1


Jun  9 16:35:44 kc3004 kernel: [  713.970342] asterisk[5122]: 
segfault at 7fedcb450716 ip 7fedeff89c8a sp 7fff4b4d5d38 error 7 in 
libpthread-2.11.1.so[7fedeff7f000+17000]

No change in config or other settings.

Any suggestions are very much welcome...
(only thing that puzzles me is that the repo contains 1.8.3 for main
asterisk, and 1.8.4 for the rest)


Hans


--
_
-- 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] Permanent restart after upgrade

2011-06-09 Thread Hans Witvliet
On Fri, 2011-06-10 at 07:21 +0800, Larry Moore wrote:
 On 10/06/2011 5:32 AM, Hans Witvliet wrote:
  Hi all,
 
  I got three asterisk-machines, two of them acting as proxies.
  On one machine (sles11sp1) i got iritating messages about not bing able
  to find codec's and other stuff, so i thought it might be time for an
  update: Stupid!
 
  I went originally from a almost working machine running:
  asterisk180-alsa-1.8.3.2-87.1
  asterisk180-dahdi-1.8.3.2-87.1
  asterisk180-1.8.3.2-87.1
  asterisk180-odbc-1.8.3.2-87.1
 
  To a machine that continuously restarts asterisk (+core dumps) running:
  asterisk180-alsa-1.8.4.2-90.1
  asterisk180-dahdi-1.8.4.2-90.1
  asterisk180-1.8.3-85.2
  asterisk180-odbc-1.8.4.2-90.1
 
 
  Jun  9 16:35:44 kc3004 kernel: [  713.970342] asterisk[5122]:
  segfault at 7fedcb450716 ip 7fedeff89c8a sp 7fff4b4d5d38 error 7 in 
  libpthread-2.11.1.so[7fedeff7f000+17000]
 
  No change in config or other settings.
 
  Any suggestions are very much welcome...
  (only thing that puzzles me is that the repo contains 1.8.3 for main
  asterisk, and 1.8.4 for the rest)
 
 
 
 One thought comes to mind, I'm not sure if you are using the same 
 computer with the same IP address or if you have set up a different 
 computer with the new installation which probably has a different IP 
 address, if the latter I would suggest you check your configuration file 
 for bindings to specific IP addresses and make sure they match the new 
 machine.
 
 Larry.

Hi Larry,

No, identical machine,
just did a zypper up -y
Because of a kernel patch (main reason for updating), i had to reboot.
Just a sip and IAX machine, no harware involved.

hw


--
_
-- 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] Permanent restart after upgrade

2011-06-09 Thread Hans Witvliet
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
 On Thu, 9 Jun 2011, Hans Witvliet wrote:
 
  I went originally from a almost working machine running: 
  asterisk180-1.8.3.2-87.1
 
  To a machine that continuously restarts asterisk (+core dumps) running: 
  asterisk180-1.8.3-85.2
 
 Any chance you have a mix of Asterisk and module versions? Was 
 Zaptel/Dahdi compiled with the proper set of headers for your kernel?
 
 Can you start Asterisk from the command line instead of the usual startup 
 script? What do the first couple of errors look like? Capturing the output 
 via the 'script' command will help.
 
 For example*,
 
   script foo
   sudo /usr/sbin/asterisk -C /etc/asterisk/asterisk.conf\
   -c -d -d -d -f -g -n -p -q -v -v -v
   exit
 
 Can you turn off auto module loading and start with no modules?
 
 *) I'm a 1.2 Luddite, so the command line arguments may have changed...
 
No dahdi/zaptel involved.
I'll be off to work in a while, report back later.


hw

--
_
-- 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] half sip registration at 1.8.3

2011-06-06 Thread Hans Witvliet
Hi all,

I've got something strange, that got me searching for quite awhile.
Configuration as followed:
Linphone on a laptop, that is connected via openvpn to a proxy.
That proxy is connected with iax to another asterisk.
On the second one i have several hard and softphones.

Behaviour at first glance:
From the softphone i can allways set up a connection,
But the otherway round fails 9 out-of 10 times.

However, if i stop-and-start linphone, the connections is allways
succesful.

First conclusion was, that if i got a diffrent (dynamic) ip-adress from
openvpn, i got to restart linphone, to force a re-registration.
Sounds reasonable, but why is linphone able to place calls, but not able
to accept them? (guests are off) I mean, if the phone is registered with
different values, also the outgoing call should fail. Not?


To avoid this behaviour, should i drastically drop the registration
duration at the softphone side? I still uses the default one (3600s).

Or should i tweak the min/max/default expiry-timers at asterisk?
Currently they are (also the default) 60/3600/120 seconds.



Hans

ps these are the lines from the console:
-- Executing [0277611@from_iax:1] noop(IAX2/kc3004-6511,
,0277611)
-- Executing [0277611@from_iax:2] answer(IAX2/kc3004-6511, )
-- Executing [0277611@from_iax:3] dial(IAX2/kc3004-6511, 

SIP/0277611 ) [Jun  6 19:03:32] WARNING[23015]: app_dial.c:2039
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)

-- Executing [0277611@from_iax:4] hangup(IAX2/kc3004-6511, )
== Spawn extension (from_iax, 0277611, 4) exited non-zero on
'IAX2/kc3004-6511'
-- Hungup 'IAX2/kc3004-6511'

corresponding lines from the ARA-dialplan:
| 118 | from_iax | 0212676 |1 | noop   | ${CALLERID},${EXTEN} |
| 119 | from_iax | 0212676 |2 | answer |  |
| 120 | from_iax | 0212676 |3 | dial   | SIP/0212676  |
| 121 | from_iax | 0212676 |4 | hangup |  |

--
_
-- 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] benefits of asterisk 1.8

2011-06-03 Thread Hans Witvliet
On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote:
 Are you suggesting that there are no bugs in 1.4 or 1.6?

I presume that you are aware of the fact that it is impossible to prove
the absence of bugs in any piece of software
You might not have detected them yet.
Furthermore behaviour that might have been coded on purpose, can be
considered eroneously some time later.

 Currently there seems to be a fear of 1.8. We're about to put it into
 production and yes, we've had issues with it, mostly due to the fact we
 use RealTime, but before you change anything it is always advisable to
 test the hell out of it.
 
 To anyone who is thinking of moving to 1.8 the question is not, 'is it
 stable?'. The question is, 'have I comprehensively tested it to show
 that it is suitable for my needs?'

If you put it into production, test at least the functions that you are
going to use. There might (and probably will) problems in the code, but
as long as it does not bother you, so what?

And don't stop testing after you put it into production: have a shadow
system (with representative configuration).
According to Murphy, side-effects will probably rise to the survice
after going into production
End-users will come up with situations you never enticipated in your
worst nightmares.


--
_
-- 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] asterisk fails when DNS or internet fails

2011-05-31 Thread Hans Witvliet
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote:
 
 On Mon, 30 May 2011, Sherwood McGowan wrote:
 
  True, but with all due respect, if the cache's TTL expires and the OP's 
  PBX cannot reach an external DNS server, they have bigger problems ;-)
  
  Slainte all!
  The Mick
 
 
 I couldn't disagree more.  In fact I think this problem is more serious 
 than it is getting credit for, when asterisk is in use in places where 
 Internet connectivity is far from stable.  I have several hotels that have 
 gone without Internet connectivity for days, and somewhere between one and 
 three days down they can only spottily call within the system, and can't 
 make outbound calls on their voice T1.  Its certainly true that they were 
 suffering without Internet access, but it is very hard to explain to the 
 owners why they can't use their phones.  In fact the symptoms are very 
 strange - inbound calls on the T1 get the auto-attendant, but internal 
 transfers fail.  No one can call outbound, and only *sometimes* do 
 internal extension to extension calls fail.
 
 I still scratch my head about what exactly asterisk is trying to lookup 
 that keeps it from being able to place internal SIP calls from extension 
 to extension, and sadly the few times this has occurred I wasn't around to 
 debug.
 
 Hasn't anyone managed to solve this with something better than a caching 
 DNS server, which seems to only last a short while?  What exactly is going 
 on that is failing?
 

What kind of info is it about?
If it is the hostname of _local_ machines/clients, you should be
authoritive. That should keep asterisk happy.
If it is about remote nodes, well if your isp-connection is lost, you
can not contact them anyway ;-(

So run locally your bind-server, authoritive for your own addresses, and
caching for external ones.

hw

--
_
-- 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] asterisk fails when DNS or internet fails

2011-05-31 Thread Hans Witvliet
On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote:
 As far as I can tell it is trying to do a reverse lookup on the IPs 
 configured on the system.  With the internet down, does the command host 
 10.10.10.1 (or whatever IPs you have on the system) take a while to come 
 back?  Unless you can do a reverse lookup of all the IPs on the system don't 
 expect Asterisk to be able to.   If your /etc/hosts is set up correct, you 
 should be able to look up any IP configured on any interface on the system 
 without delay.
 
 I'm sure there are other places Asterisk tries to do DNS lookups, but the 
 above info has solved this issue for me in the past.
 

I'm not sure if that's all is true.
Sure, if you add a line in /etc/hosts, that works for most applications,
as not all commands follow /etc/resolv.conf

i just tried, adding a line to /etc/hosts.
ping hostname works, but host hostname fails, just as host ip-address.
So even when you only put ip-addresses (brrr) into your config files,
the reversed-lookup will still spoil the party.


--
_
-- 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] Why PRI not BRI ?

2011-05-30 Thread Hans Witvliet
On Mon, 2011-05-30 at 13:57 +0530, virendra bhati wrote:
 Thanks a lot all, 
 Now my view is clear ...
 
 On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson gordon
 +aster...@drogon.net wrote:
 On Sun, 29 May 2011, virendra bhati wrote:
 
 Hi List,
 
 I have stupid question but I want to know it. Why we
 use the PRI insted of
 BRI ? Just for the sake of number of lines or any
 thing else ?
 
 
 It probably depends on your country.
 
 In mainland europe (or maybe just Germany), ISDN2e (BRI) is
 very popular - not uncommon in home installations too.
 
 In the UK, it's almost the standard in small businesses - the
 migration path seems to be from a single line to 3 lines
 sharing the same number to ISDN2e...
 
 There was a push in the UK to support BRI in the home (~10
 years back, under the name Home Highway), but it came at a
 time when ADSL was almost upon us, and BT in their infinite
 wisdom removed a lot of the ISDN features that make it
 actually useful...
 
 I don't think BRI ever caught on in the US - It was analogue
 or PRI (or channelised/fractional T1 or whatever it's called)
 Probably made it much easier for the telcos to support (and
 afford)

Only reason for using bri instead of pri in the number of  voice chanels
and costs. It took ages  before telco's realised that with fractured-E1
they could save a lot of costs (telco/customers) while offering a cheap
upgrade path. At that time that ISDN was introduced, the costs in
installing a pri-interface in the local-exchange was identical to
installing a bri-interface.

Only reason nowadays for using bri instead of pots, is that you get the
incoming speech channel already digitialised.


 And why SIP is used for making calls rather then IAX?
 Even we know IAX takes
 1 channel for making calls?
 
 
 SIP is an open standard that's been around since the late
 90's. IAX, which is also open and free was only just accepted
 as a standard last year, but even so, there's inertia. Very
 few phone manufacturers are using it - why should they, when
 they've been using SIP for years, and the same PBX that works
 with IAX also works with SIP... (And does any other PBX
 support IAX yet?)
 

Freepbx is the only other afaicr.
Only a limited number of clients.

hw

--
_
-- 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] [Fwd: FW: realtime mysql - p4]

2011-05-22 Thread Hans Witvliet
On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote:
 Post your cdr_mysql.conf and res_mysql.conf and we'll take it from
 there.
 
 Don't forget to remove any 'private' info first (like passwords).
 
 Cheers

Tnx for the offer,
Wil get the files when got back at the office.
I presume that cdr_mysql.conf is only relevant for storing
call-data-records? Perhaps that is something for later on.

For now, i have to show a working *, with all sip-details in a mysql-DB.
Other people pointed out that other means (postgres, ldap) might work
better, but that's not an option for me.

Hans

--
_
-- 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] [Fwd: FW: realtime mysql - p4]

2011-05-19 Thread Hans Witvliet
Ok, i tried the suggestion:
Instead of:
sippuser = resource, database_name, table_name
sippeer  = resource, database_name, table_name
 
I put in:
sippuser = resource, context, table_name
sippeer  = resource, context, table_name

Unfortunately, with the same results.
btw i tried both general as default

Besids the commands i tried below, isn't there any other way to see what's 
going on?

Perhaps it is totally unrelated, but if i perform a mysql-login on the prompt,
i first have to select the database manualy, ie it isn't selected by default 
for the created mysqluser
[in this case: voipadmin]

Other wild idea, is there a minimum number of fields that haved to be filled?

And why is asterisk complaining about not being able to find the databse, when 
trying to fill it from the asterisk-CLI?
My database _is_ named asterisk..
 kc3054*CLI  realtime update sipusers set SET port = 4343 WHERE name =
 0277611 Failed to update. Check the debug log for possible SQL
 related entries.
 Command 'realtime update sipusers set SET port = 4343 WHERE name =
 0277611' failed.
 [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql:
 MySQL RealTime: Invalid database specified: 'asterisk' (check
res_mysql.conf)

I mean, is that silly or what?


 
 
 # grep mysql extconfig.conf |grep sip
 ;sipusers = mysql,asterisk,sip_devices
 ;sippeers = mysql,asterisk,sip_devices
 ;sipusers = mysql,general,sip_devices
 ;sippeers = mysql,general,sip_devices
 sipusers = mysql,default,sip_devices
 sippeers = mysql,default,sip_devices
 
 
 kc3054*CLI module show like mysql
 Module Description  Use 
 Count 
 cdr_mysql.so   MySQL CDR Backend0 
 
 res_config_mysql.soMySQL RealTime Configuration Driver  0 
 
 app_mysql.so   Simple Mysql Interface   0 
 
 3 modules loaded
 kc3054*CLI
 kc3054*CLI sip show users
 Username   Secret   Accountcode  Def.Context  
 ACL  ForcerPort
 j.witvliet geheimdefault  
 No   Yes   
 027761125b06d3a0b5ef73   default  
 No   Yes   
 kc3054*CLI
 kc3054*CLI sip show peers
 Name/username  HostDyn 
 Forcerport ACL Port Status Realtime
 0277611(Unspecified)D   N 
  0Unmonitored 
 j.witvliet (Unspecified)D   N 
  0Unmonitored 
 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] 
 kc3054*CLI kc3054*CLI 
 
 kc3054*CLI
 kc3054*CLI realtime mysql cache
 kc3054*CLI realtime mysql status
 general connected to asterisk@127.0.0.1, port 3306 with username voipadmin 
 for 18 seconds.
 kc3054*CLI 
 


--
_
-- 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] asterisk18 - realtime/mysql - take 3

2011-05-18 Thread Hans Witvliet
Still a couple of questions..

I did configure extconfig.conf
...
;iaxusers = odbc,asterisk
;iaxpeers = odbc,asterisk
;sipusers = odbc,asterisk
sipusers = mysql,asterisk,sip_devices
sippeers = mysql,asterisk,sip_devices
;sippeers = odbc,asterisk
;sipregs = odbc,asterisk
;voicemail = odbc,asterisk
;extensions = odbc,asterisk
;meetme = mysql,general
;queues = odbc,asterisk
;queue_members = odbc,asterisk
;musiconhold = mysql,general
;queue_log = mysql,general

So only defining sipusers  sippeers for mysql


And noticed two files for configuring mysql-stuff:
file: res_config_mysql.conf
database access config: host, user, pwd

file: res_odbc.conf
in section [mysql2]: mysql database config: host, user, pwd

So, i configured them both...

Quick check:kc3054*CLI sip show users
Username Secret Accountcode  Def.Context  ACL  ForcerPort
j.witvliet geheim  default  No   Yes   
0277611 25b06d3a0b5ef73default  No   Yes   
kc3054*CLI 

kc3054*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status Realtime
0277611 (Unspecified)  D   N  0Unmonitored 
j.witvliet (Unspecified)   D   N  0Unmonitored 
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2
offline] kc3054*CLI 


Here i see at for both users and peers ONLY the statis entries from
sip.conf file, nothing from mysql...


kc3054*CLI realtime mysql status
general connected to asterisk@127.0.0.1, port 3306 with username
voipadmin for 5 seconds.
kc3054*CLI 
=No warnings/errors but nothing else either...

kc3054*CLI
kc3054*CLI realtime mysql cache
kc3054*CLI 
=No warnings/errors but nothing else either...

the module res_config_mysql.so is loaded,


Try todo something else:
kc3054*CLI realtime update sipusers set SET port = 4343 WHERE name =
0277611 Failed to update. Check the debug log for possible SQL
related entries.
Command 'realtime update sipusers set SET port = 4343 WHERE name =
0277611' failed.
[May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql:
MySQL RealTime: Invalid database specified: 'asterisk' (check
res_mysql.conf) kc3054*CLI 

== here the system talkes about _another_ config file! ==
So which file should i configure:
A) res_config_mysql.conf
B) res_odbc.conf
C) res_mysql.conf
But even when i put my credentials in all three of them, still no show!

DB check:
mysql -h localhost -u voipadmin -p
Enter password: 
Server version: 5.0.67 SUSE MySQL RPM
mysql use asterisk; select name,username,secret,host,nat from
sip_devices;
Database changed
+-+-++-+-+
| name| username| secret | host| nat |
+-+-++-+-+
| 0031756 | 0031756 | geheim | dynamic | Yes |
+-+-++-+-+
1 row in set (0.00 sec)
mysql 

According to *,TDG, page 349:
Also filled the file /etc/unixODBC/odbcinst.ini, and the command
odbcinst -q -d produced the required result: [MySQL]
I presume i made a silly mistake/omission, but i fail to see how i can
detect that, or other steps to test the correct configuration of ARA.
So it looks that i'm stuck.

Can not imagine that i'm  the first here!
But even from the definitive guide, chapter 16 and onwards, it isn't
clear if you should use the mysql-stuff directly of through the
odbc-routines

Kind regards, Hans


--
_
-- 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] iptables for Asterisk - Any good guides out there?

2011-05-14 Thread Hans Witvliet
On Sat, 2011-05-14 at 19:51 -0400, Bruce B wrote:
 Hi everyone,
 
 
 I want to issue the command:
 
 
 iptables -F
 
 
 and then rebuild everything from the beginning with a very limited
 scope and then without locking myself block all other traffic. Can you
 suggest what I should put in the shell that would get me this:
 
 
 Allow traffic from subnet 172.16.0.0/24  (my VPN tunnels) - All
 traffic including those of Asterisk and HTTP - I trust this network
 Allow traffic from subnet 192.168.1.0/24(other side of VPN
 network) - All traffic including those of Asterisk and HTTP - I trust
 this network
 Allow traffic from single IP of DID provider - 5060 TCP/UDP and
 1-10200 UDP
 Allow VPN access on port 1194 UDP   --- I have that figured out to be
 (iptables -A INPUT -p udp -m udp --dport 1194 -j ACCEPT) works for
 this.
 
 
 BLOCK all other traffic - Important most of all
 
 
 Please note that from the subnets I want to allow every single port
 possible and all traffic. I specially have problems with getting a
 whole subnet be able to access everything.
 
 
 Thanks

It's a bit more complicated

Firstly you have to set the default rules FIRST
$IPT -P INPUT DROP
$IPT -P OUTPUT ACCEPT
$IPT -P FORWARD ACCEPT
And then do the flusing, not the otherway round
After that you can add rules to accept trafic

after the last rules, it is handy to put:
$iptables -A INPUT  -i $EXTERNAL_DEV -j LOG --log-prefix  EXT; INC 
iptables -A OUTPUT  -o $EXTERNAL_DEV -j LOG --log-prefix  EXT; OUT 
iptables -A FORWARD -i $EXTERNAL_DEV -j LOG --log-prefix  EXT; FWD 
So can can see in the syslog what you are missing ;-)



I'll guess, you would also like to accepts ntp,dhcp, domain-dns from
your isp-provider.

Perhaps also http, https, pop, pops, imap, imaps.
And probably some more, depending on your need
So'll see them soon enough in your logfiles

hw

--
_
-- 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] Realtime - ara180 - part-2

2011-05-13 Thread Hans Witvliet
On Thu, 2011-05-12 at 21:30 +0200, bakko wrote:
 Hi,
 
 look if you have res_config_mysql.so module instaled on your asterisk.
 
 On CentOS /usr/lib/asterisk/modules
 
 Regards
Tnx for your reply.
It turned out, that mysql-support was in a different rpm (addons)
As systems are never connected to the big-bad-external world, it took
some time to do an upgrade and fetch the missing rpm. had some serious
complaining l-users to tend to ;-)

So i left the config unchanged, but noticed:
kc3054*CLI sip show users
Username  Secret  Accountcode Def.Context  ACL  ForcerPort
j.witvliet geheim  default  No   Yes   
0277611 25b06d3a0b5ef73default  No   Yes   
kc3054*CLI 


kc3054*CLI sip show peers
Name/username Host  Dyn Forcerport ACL Port Status Realtime
277611  Unspecified)  D   N  0Unmonitored 
j.witvliet  (Unspecified) D   N  0Unmonitored 
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2
offline] kc3054*CLI 


kc3054*CLI realtime mysql status
general connected to asterisk@127.0.0.1, port 3306 with username
voipadmin for 2 hours, 7 minutes.
kc3054*CLI 

kc3054*CLI
kc3054*CLI  realtime mysql cache
kc3054*CLI 


So, no more complaints at the cli or logfile,
But the sip-entry from mysql 0031756 still does not show up.

I hope that any suggestions/help will be educational to others

Hans

--
_
-- 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] Realtime - ara180

2011-05-12 Thread Hans Witvliet
Hi all,

A week or so down the list, i read that not many people were using
realtime on an Asterisk18, so i had this afternoon a go at it...
[sorry for the inconveneant line-wraps]

First i did:
mysql create database asterisk;
mysql grant all on asterisk.* to 'voipadmin'@'localhost' identified by 

next i used the info from the wiki:

CREATE TABLE `sip_devices` (
 `id` int(11) NOT NULL AUTO_INCREMENT,
 `name` varchar(80) NOT NULL DEFAULT '',  `context` varchar(80) DEFAULT
NULL, 

[and lots more]

and populated it with a test user:
mysql 'secret'; insert into sip_devices (name, secret, username, host,
nat) values ('0031756', 'geheim', '0031756', 'dynamic', 'Yes');

tested the database:
mysql -h localhost -u voipadmin -p
Enter password: 
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 12645
Server version: 5.0.67 SUSE MySQL RPM
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
mysql select * from sip_devices\G
ERROR 1046 (3D000): No database selected
mysql use asterisk;
Database changed
mysql select name,username,secret,host,nat from sip_devices;
+-+-++-+-+
| name| username| secret | host| nat |
+-+-++-+-+
| 0031756 | 0031756 | geheim | dynamic | Yes |
+-+-++-+-+
1 row in set (0.00 sec)
mysql 

So-far all looked, as expected, great


Changed the *-config files:
(method, db-name, table-name)
;
;iaxusers = odbc,asterisk
;iaxpeers = odbc,asterisk
;sipusers = odbc,asterisk
sipusers = mysql,asterisk,sip_devices
sippeers = mysql,asterisk,sip_devices
;sippeers = odbc,asterisk
;sipregs = odbc,asterisk
;voicemail = odbc,asterisk

And restarted the asterisk-process...
Some lines from /var/log/asterisk/messages:
May 12 14:05:33] WARNING[2585] config.c: Realtime mapping for 'sippeers'
found to engine 'mysql', but the engine is not available 
[May 12 14:05:33] WARNING[2585] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available 
[May 12 14:05:55] WARNING[2630] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available


And on the asterisk console:
kc3054*CLI sip show peers
Name/username  Host  Dyn Forcerport ACL Port Status 
0277611  (Unspecified)   D   N  0Unmonitored 
j.witvliet   (Unspecified)   D   N  0Unmonitored 
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2
offline] [May 12 14:17:47] WARNING[2630]: config.c:2045 find_engine:
Realtime mapping for 'sippeers' found to engine 'mysql', but the engine
is not available

CLI sip show users
Username Secret Accountcode  Def.Context  ACL  ForcerPort
j.witvliet  geheim  default  No   Yes   
0277611 25b06d3a0b5ef73 default  No   Yes   
CLI 



1) as shown above, access to mysql seems to be OK,
2) * did not complain at sip-show-users but only for sip-show-peers
fyi, i use: asterisk180-1.8.3.2-87.1.x86_64.rpm

Any suggesions?

Hans

--
_
-- 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] [SOT] Virtualising Asterisk

2011-05-08 Thread Hans Witvliet
On Sat, 2011-05-07 at 16:24 +0100, --[ UxBoD ]-- wrote:
 I know a lot has changed over the past couple of years, and even
 monthly, and that Asterisk running within a virtualised environment is
 very happy indeed. If one would only be using SIP/IAX would Xen/KVM be
 the best solution ? / or perhaps VServer/LXC maybe advantageous due to
 binary hashing.  Your thoughts would be very welcome.
 -- 

At work we have a couple of asterisk instances.
Some of them are used for doing interfaceing job, like towards isdn-BA
or isdn-PRA, skype or GSM. (those do only _that_ and nothing more.

Others (dialplan, voicemail, etc etc) are running as an XEN-image.
Reason: much more easier to replace
Some of the are still 1.4 while others run several different 1.6
versions.
Currently i'm experimenting with a bunch of virtual 1.8-systems.

from my point of view sles/xen works rock solid.


hw


--
_
-- 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] Discussion: Are we ready to leave 1.4 behind?

2011-05-02 Thread Hans Witvliet
On Wed, 2011-04-27 at 21:34 +0200, Olle E. Johansson wrote:
 Friends,
 
 We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
 According to the release plans, support for 1.4 was scheduled to close in 
 April 2011 - basically now. 
 After that, only security patches would be committed. This is already a delay 
 from the original plan published by Russell Bryant.
 
 Unfortunately, I think this is way too early. 
 My feeling and experience is that 1.8 is not ready for production in the 
 environments I work in - large scale installations. 
 Customers are not planning migration and all new installs are still 1.4.
 Tests we've been doing with 1.8 has failed within just a short time and so 
 badly that customers has not paid me to spend any further time with 1.8.
 

Just a thought
If Digium / the community realy want an objective way of deciding
whether can/should migrate to any other version, you realy need a
feature-matrix (pethaps starting from version 1.2.*)

And for every and each version a statement if it is:
- discontinued
- tested
- test finalized, result indicating it is fully and identically
functional
- test finalized, result indicating that this feature is changed in
either behaviour of configuration
- not yet tested.

I realize it is quite a job to do, but if done it would be for everyone
easily to see if it is worthwhile to start migrating.

Anyway for both documentation purposes and bugtracking it would be nice
if each and every feature has a unique numerique identifier.

And perhaps there is a fair chance that the people from the quality
department at Digium already have such a list.


hw

--
_
-- 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


  1   2   3   >