Re: [asterisk-users] Record() Cmd and My SQL

[Govind, Mahesh (NSN - IN/Bangalore)]

The reason is when doing a load balancing  , We  cannot confine the
recording to a particular asterisk machine ( If we have more than one
asterisk machine in the topology ).

So a centralized mechanism might be better . So that any machine can
access the recording .
Regards
Mahesh


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext David
Backeberg
Sent: Thursday, September 23, 2010 9:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd and My SQL

On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
 wrote:
> HI ,
>
> Is there Any way is there so that I can store my recordings directly
to a
> database rather storing the same to a file .

Please, please, please tell us why you would want to do that.

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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)

[bruce bruce]

Thanks for the detailed info. Problem was solved by including Server B
subnet as the localnet of the Server A (OpenVPN server) and setting each
extension NAT=NO.

Your points are good guides for future problem diagnoses.

Thanks again,
Bruce

On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt  wrote:

>
> > I don't think it's an endpoint issue. I think the SIP packet headers get
> > over-written by the tunnel (openvpn) protocol.
>
> I'd be rather astonished if OpenVPN itself were responsible for this.
> As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
> of any sort.  It just provides the "bit pipe" through the tunnel.
>
> I'd suggest several other possible culprits:
>
> (1) Server B might be doing higher-level protocol rewriting (i.e.
>SIP border gateway stuff) prior to routing the SIP packets
>through the OpenVPN tunnel.
>
>This might happen if Server B were configured to use the
>Linux "iptables" features, with a SIP protocol module and
>Network Address Translation features.
>
>The fix would be to disable NAT and boundary processing in
>Server B's routing functions.
>
> (2) The SIP endpoints (phones) might be configured to "discover
>their external address", via STUN or a similar mechanism.
>
>The fix would be to change the endpoint device configuration.
>
> I think you'll need to use Wireshark or a similar sniffer, to see
> what the SIP traffic looks like at several points along the path,
> so you can locate the earliest point at which the wrong address is
> in the SIP packet payload.
>
> Several examination points come to mind:
>
> -  Right after the packet leaves the endpoint device.  I'd suggest
>   using a laptop running Wireshark as a passive packet sniffer...
>   connect the endpoint device and the laptop to an Ethernet hub
>   (not a switch!) and sniff the packets before any router gets
>   its hands on them.
>
> -  As the packets enter Server B - use Wireshark on Server B and
>   have it tap into the incoming Ethernet interface.
>
> -  As the packets are pushed out of Server B's routing layer into
>   the OpenVPN tunnel.  Use Wireshark to monitor the "tap" or
>   "tun" virtual interface, to which the kernel transmits the packets
>   that OpenVPN is to convey.
>
> -  As the packets come out of the tap/tun device on Server A.
>
> In scenario (1) I described above, you'd see the packets be correct
> at the first and second Wireshark sniffing points, and incorrect at the
> third and fourth (i.e. the modifications are being performed in
> Server B's routing/NAT'ing layer).
>
> In Scenario (2), they'd be incorrect at every point, including just
> after they come out of the IP-phone.
>
> In the scenario you described, they'd be correct at the first, second,
> and third points, and wrong at the fourth.
>
>
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[asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address

[Mike]

Hi,

 

I have a server with multiple IP address, Asterisk binding with all of them.
I'd like Asterisk to reply to a SIP peer from the same IP address as the
peer used to register to Asterisk (as opposed to using the main IP address
all the time regardless of how the peer communicated with Asterisk).

 

Is this possible? I know it wasn't with 1.4, but I was told 1.6 had
something like this (something to do with not breaking SIP over TCP)

 

Mike

 

 

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[asterisk-users] rtp problem with 1.8.0-rdc1

[covici]

Hi.  I am having a very strange problem --aren't they all -- with the
release candidate.  I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic ceases!  Now, a version which does work is
r281875, this does not happen in that vrsion, but right after that this
strange thing starts and is not fixed in the current one.

Any assistance here would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] realm: security issue

[Tarek Sawah]

Bilal,
If you are using 3G or Wifi with your Nokia Native SIP Client.. try to
connect via an internet connection sharing machine.. it seems that your ISP
is blocking INBOUND SIP packets.
Test and let me know

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, September 23, 2010 11:24 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] realm: security issue

No, I do not think that my provider blocked my IP address, because I am able
to register for the Asterisk (at that IP address) from an IP Phone, but not
from the mobile. It is well known that the mobile use the digest
authentication (realm) which is not used in the IP Phone.

Any advise?
 
> >From what you explained it seems to me that your mobile
> provider has blocked
> your sip communication altogether. Have you tried changing
> IP address of
> your asterisk server? If changing IP works, then probably
> your provider has
> blocked you sip communication by IP only.
> 
> Zeeshan A Zakaria
> 
> --
> www.ilovetovoip.com
> 
> On 2010-09-23 7:22 AM, "bilal ghayyad" 
> wrote:
> 
> Hi All;
> 
> I have my friend that use his mobile (Nimbuz) to connect
> for the Asterisk
> and his account was working fine. Suddenly it stop working
> (not able to
> register).
> 
> >From my mobile (Nokia) I was able to register using my
> username and
> password, so I tried to register using his (my friend)
> username and password
> (that was using them from Nimbuz), it did not work. I come
> back trying to
> register using my origin username and password (which was
> working fine just
> before a while), it did not work. I removed my username and
> my friend
> username from the Asterisk and then I created a new
> username and password
> (different than all other) and I tried to register from my
> mobile, also it
> did not work !!!
> 
> I start beleive that it is something related to detecting a
> hacking (maybe
> Nimbuz does not use a good security), this caused the MAC
> to be considered
> as hacked.
> 
> Please, can someone advise me how to resolve this problem?
> Where I can find
> those MACs that need to be removed from block list? What
> can I do to get out
> from this problem?
> 
> Any advise?
> Regards
> Bilal


  

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Re: [asterisk-users] realm: security issue

[bilal ghayyad]

No, I do not think that my provider blocked my IP address, because I am able to 
register for the Asterisk (at that IP address) from an IP Phone, but not from 
the mobile. It is well known that the mobile use the digest authentication 
(realm) which is not used in the IP Phone.

Any advise?
 
> >From what you explained it seems to me that your mobile
> provider has blocked
> your sip communication altogether. Have you tried changing
> IP address of
> your asterisk server? If changing IP works, then probably
> your provider has
> blocked you sip communication by IP only.
> 
> Zeeshan A Zakaria
> 
> --
> www.ilovetovoip.com
> 
> On 2010-09-23 7:22 AM, "bilal ghayyad" 
> wrote:
> 
> Hi All;
> 
> I have my friend that use his mobile (Nimbuz) to connect
> for the Asterisk
> and his account was working fine. Suddenly it stop working
> (not able to
> register).
> 
> >From my mobile (Nokia) I was able to register using my
> username and
> password, so I tried to register using his (my friend)
> username and password
> (that was using them from Nimbuz), it did not work. I come
> back trying to
> register using my origin username and password (which was
> working fine just
> before a while), it did not work. I removed my username and
> my friend
> username from the Asterisk and then I created a new
> username and password
> (different than all other) and I tried to register from my
> mobile, also it
> did not work !!!
> 
> I start beleive that it is something related to detecting a
> hacking (maybe
> Nimbuz does not use a good security), this caused the MAC
> to be considered
> as hacked.
> 
> Please, can someone advise me how to resolve this problem?
> Where I can find
> those MACs that need to be removed from block list? What
> can I do to get out
> from this problem?
> 
> Any advise?
> Regards
> Bilal


  

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[asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.

  * Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
 Review: https://reviewboard.asterisk.org/r/938/)

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the last beta release as reported by
the community. A sampling of the changes in this release candidate include:

  * Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)

  * Fixes a bug in manager.c where the default configuration values weren't 
reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)

  * Various fixes for the calendar modules.
(Patched by Jan Kalab.
 Reviewboard: https://reviewboard.asterisk.org/r/880/
 Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
 Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)

  * Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)

  * Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn't restart.
(Closes issue #17408, Reported, patched by sysreq)

  * Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)

  * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 5%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)

  * Don't clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when 
a
registration expiries so realtime probably shouldn't either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
 mnicholson)

  * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
 twilson)

  * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!

A short list of available features includes:

  * Secure RTP
  * IPv6 Support in the SIP channel driver
  * Connected Party Identification Support
  * Calendaring Integration
  * A new call logging system, Channel Event Logging (CEL)
  * Distributed Device State using Jabber/XMPP PubSub
  * Call Completion Supplementary Services support
  * Advice of Charge support
  * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2

Thank you for your continued support of Asterisk!

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[asterisk-users] Unable to make outgoing call on E1

[Adolphe Cher-aime]

Hello everyone,

 I'm using redfone fonebridge to have Pstn  
connectivity on my asterisk box.

I can receive in coming calls however outgoing calls don't go to  
provider. It's seems it's a span config problem. Because in systemconf  
when I try to config span as follow
  span=1,0,2,ccs,hdb3,crc4

And start dhadi I have the following message

Unable to config ... Bad argument(22).


Please any help


Adolphe Cher-aime
 From my Iphone

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Re: [asterisk-users] Asterisk- speech to text(Voicemail totext message)

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Sherrill
Sent: Thursday, September 23, 2010 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk- speech to text(Voicemail totext
message)



>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On 
>Behalf Of Danny Nicholas
>Sent: Wednesday, September 22, 2010 5:04 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Asterisk- speech to text(Voicemail to text
message)
>
>FWIW, the current state of Speech-to-text will let you do a 70-95% accurate
translation of 
>incoming voicemails depending on clarity/dialect/training.  Also depends on
language of 
>"native" speakers.  For 100% reliability, this still requires Human
intervention.

I'd like to do this too.  Poking around, it looks like res_speech.so is the
library to enable it, but an actual separate program to convert from voice
to text is needed, like Sphinx or VXI?  I haven't found anything yet that
describes how to connect it to voicemail.  Examples are welcome, if anyone
has one to point at/paste.

Looking at Sphinx and the available documentation, I think these things to
be true.
#1 - res_speech.so isn't necessary since Sphinx operates as a external
module as opposed to the resident modules of Vestec and Lumenvox.
#2 - Didn't really find a good "on-the-fly" example of processing the file
as it came in.


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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)

[Dave Platt]

> I don't think it's an endpoint issue. I think the SIP packet headers get
> over-written by the tunnel (openvpn) protocol.

I'd be rather astonished if OpenVPN itself were responsible for this.
As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
of any sort.  It just provides the "bit pipe" through the tunnel.

I'd suggest several other possible culprits:

(1) Server B might be doing higher-level protocol rewriting (i.e.
SIP border gateway stuff) prior to routing the SIP packets
through the OpenVPN tunnel.

This might happen if Server B were configured to use the
Linux "iptables" features, with a SIP protocol module and
Network Address Translation features.

The fix would be to disable NAT and boundary processing in
Server B's routing functions.

(2) The SIP endpoints (phones) might be configured to "discover
their external address", via STUN or a similar mechanism.

The fix would be to change the endpoint device configuration.

I think you'll need to use Wireshark or a similar sniffer, to see
what the SIP traffic looks like at several points along the path,
so you can locate the earliest point at which the wrong address is
in the SIP packet payload.

Several examination points come to mind:

-  Right after the packet leaves the endpoint device.  I'd suggest
   using a laptop running Wireshark as a passive packet sniffer...
   connect the endpoint device and the laptop to an Ethernet hub
   (not a switch!) and sniff the packets before any router gets
   its hands on them.

-  As the packets enter Server B - use Wireshark on Server B and
   have it tap into the incoming Ethernet interface.

-  As the packets are pushed out of Server B's routing layer into
   the OpenVPN tunnel.  Use Wireshark to monitor the "tap" or
   "tun" virtual interface, to which the kernel transmits the packets
   that OpenVPN is to convey.

-  As the packets come out of the tap/tun device on Server A.

In scenario (1) I described above, you'd see the packets be correct
at the first and second Wireshark sniffing points, and incorrect at the
third and fourth (i.e. the modifications are being performed in
Server B's routing/NAT'ing layer).

In Scenario (2), they'd be incorrect at every point, including just
after they come out of the IP-phone.

In the scenario you described, they'd be correct at the first, second,
and third points, and wrong at the fourth.


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Re: [asterisk-users] Asterisk and Digium TC400B

[Bryant Zimmerman]

On 09/23/2010 06:48 AM, Tarek Sawah wrote:
> Greetings,
> Because of the heavy load and the high expectations of an asterisk 
server
> offered as a solution integrated with our CRM software.. we were looking
> into other possibilities than software Licenses for G729 and G723 
codecs..
> to lower the pressure on the processor giving it more space to do more 
work.
> We heard of a hardware (PCI CARDS) can be used with Asterisk that does 
the
> work. And we stumbled with Digium TC400B.
> Could be a newbie's question.. but does that serve our needs? As we have 
not
> pressured a server before up to 1400 extensions with 600 outbound SIP 
calls
> (customer's needs).
> The server in question is Core I7 16 GB ram and Raid 10 SAS drives.
> We need to know how many calls with G729 or G723 can this server handle? 
And
> as far as we can see this Digium card can be a cheaper solution If
> calculating the CPU cost plus the licenses for each channel.
> One more question.. can we add two of those cards to the server? Will it 
be
> efficient?

Hi Tarek,
I have TC400B cards installed and they work fine. You get up to 120 
channels per card.
You can install multiple cards and they work good. 
The new sangoma G729 cards have the ability to do up to 2400 channels per 
card depending on the configuration purchased.
The sangoma option is really good option once you get the 120 channel 
level.

The real question is how many channels do you need to transcode. In certian 
combinations asterisk can eat g729 channels. Let's say you are coming g729 
and recording and then going back out g729 you may eat up to 4 
encode/decode license. For some calls if your end points are g729 and your 
carrier is g729 you many not need any license. It comes down to how many of 
your calls will need access to non g729 prompts and how many calls will 
need to be converted due to differing source formats. If you can convert 
your ivr prompts to g729 you get a win here. But playing voicemail files 
will never use g729 as it is not currently a supported record format as far 
as I have found

My guss is one of the sangoma 400 license cards would likely meet your 
needs That will range between $2200 and $2300
A single sangoma 240 license card is between $1550 and $1650
The digium TC400B selles for beteen $1025 and $1200.00

If you want more details you can contact me off list.

Bryant

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Re: [asterisk-users] Asterisk- speech to text(Voicemail to text message)

[Justin Sherrill]

>From: asterisk-users-boun...@lists.digium.com 
>[mailto:asterisk-users-boun...@lists.digium.com] On 
>Behalf Of Danny Nicholas
>Sent: Wednesday, September 22, 2010 5:04 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Asterisk- speech to text(Voicemail to text 
>message)
>
>FWIW, the current state of Speech-to-text will let you do a 70-95% accurate 
>translation of 
>incoming voicemails depending on clarity/dialect/training.  Also depends on 
>language of 
>"native" speakers.  For 100% reliability, this still requires Human 
>intervention.

I'd like to do this too.  Poking around, it looks like res_speech.so is the 
library to enable it, but an actual separate program to convert from voice to 
text is needed, like Sphinx or VXI?  I haven't found anything yet that 
describes how to connect it to voicemail.  Examples are welcome, if anyone has 
one to point at/paste.

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Re: [asterisk-users] Digest Username/auth name mismatch ‏

[Steve Howes]

On 23 Sep 2010, at 17:29, t. k wrote:
> >Isn't there any way to configure the username in the hardphone to be 
> >just "?
> Yes.there is no way to cofigure as "" in the hardphone.It will cost and 
> spend time a lot to implement

Then I think the short answer is that it's not compatible.

Steve
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Re: [asterisk-users] Digest Username/auth name mismatch ‏

[t. k]

Hi

 

>Isn't there any way to configure the username in the hardphone to be 
>just "?
Yes.there is no way to cofigure as "" in the hardphone.It will cost and 
spend time a lot to implement

 

thanks. 
 
> Date: Tue, 21 Sep 2010 14:04:17 +0100
> From: s...@open-t.co.uk
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Digest Username/auth name mismatch‏
> 
> 
> 
> On 09/21/2010 04:26 AM, t. k wrote:
> >
> > Hi
> >
> > Thanks for help.
> >
> >
> >> I will try to help. But others might know more. What SIP client are you
> >> using - a softphone, a hardphone? It looks like the client is sending
> >> the full " at 192.168.0.1" instead of just "" as the username.
> > Sebastian
> >
> > That's right.hardphone is sending  at 192.168.0.1 for Proprietary 
> > specification.
> > ※Digest usrname can't change with SIP Client.
> > so I would like to solve this hardphone issue with asterisk.
> 
> Isn't there any way to configure the username in the hardphone to be 
> just "?
> 
> Sebastian
> 
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[asterisk-users] Asterisk Transfer/call patching support

[Dan Cropp]

I'm coming to Asterisk from a traditional PSTN environment, so forgive
me if I use the wrong Asterisk/SIP terminology.

 

I need to make a product where calls come in go through various menus
and based on various configurations perform attended transfers, blind
transfers, and patch callers together.

 

For patching two calls together, my thought is that this would be a
conference in Asterisk.  Is this correct?

 

For attended transfers, is there a way to perform this from a dial plan?
Or would I need to use AMI?

 

Also, with Asterisk transfers (SIP and PRI calls), will the transferred
call disappear from Asterisk?  For example, with PRI QSIG transfers, if
the external switch allows it, both parties of a PSTN call are removed
from a switch and instead the parent switch becomes responsible for the
bridged calls.

 

I'm using the current Asterisk trunk with plans to use Asterisk 1.8 once
it's released.

 

Have a great day!

Dan

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Re: [asterisk-users] Record() Cmd and My SQL

[David Backeberg]

On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
 wrote:
> HI ,
>
> Is there Any way is there so that I can store my recordings directly to a
> database rather storing the same to a file .

Please, please, please tell us why you would want to do that.

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[asterisk-users] Forking a call

[Mike]

Hi,

 

Using 1.6.2.13.  

 

I'd like to know how I can force Asterisk to fork a call.  To simplify
things, Let's say I have an out context (for outbound calls) and an in (for
inbound).  If person A wants to call person B, and both are on my servers, I
don`t want to send the call out.  I want all this to happen internally on my
server.

 

The problem is if I use some condition to send calls in my out context back
to my in context, some channel variables get mixed up, and (for example)
when the calling part puts the called party on hold, the music on hold used
is the called party's music.  I am sure there are some less benign problems
that could come with that.

 

Is ForkCDR() what I am looking for? Any things I gotta watch out for when
using it?

 

I basically would like Asterisk to treat this call as two separate calls, as
if one was completely outbound and the second an independant inbound call.

 

Mike

 

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Re: [asterisk-users] Record() Cmd and My SQL

[Danny Nicholas]

 

 

 

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext Danny
Nicholas
>Sent: Thursday, September 23, 2010 7:27 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Record() Cmd and My SQL

 

>Just my .02 - 

>#1 - voip-info.org is a good resource for finding this kind of script

>#2 - I think you are asking for trouble doing a direct recording using the
PHP/MySQL combination; Personally I would do C/MySQL but the exposure you
face depends on the length of your recording(s).

>#3 - since you are going to need to go the AGI route anyway, why not do a
MixMonitor/AGI combination so you get the speed/reliability of a disk
write-to-file?

>>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Govind, Mahesh
(NSN - IN/Bangalore)
>>Sent: Thursday, September 23, 2010 9:49 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [asterisk-users] Record() Cmd and My SQL

 

>>Thanks .

>>I checked in voip info but I was not able to find a script which records
directly to database . All scripts were recording directly to files .

 

>>About #2 and  #3 can you please provide some more information .

 

>>Regards

>>Mahesh

 

Okay, I did some googling and according to this link

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

 

You can't record directly to a database.

 

If these are "short" recordings (less than 60 seconds), I would do something
like this:

 

Exten => 1234,1,playback(intromessage)

Exten => 1234,2,record("/tmp/filex.gsm")

Exten => 1234,3,MYSQL( write /tmp/filex as a blob into a record)

 

Refer to this link also

http://www.bigresource.com/MYSQL-Audio-Files-in-BLOB-egyxueFP.html

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Re: [asterisk-users] Asterisk and Digium TC400B

[Shaun Ruffell]

On 09/23/2010 06:48 AM, Tarek Sawah wrote:
> Greetings,
> Because of the heavy load and the high expectations of an asterisk server
> offered as a solution integrated with our CRM software.. we were looking
> into other possibilities than software Licenses for G729 and G723 codecs..
> to lower the pressure on the processor giving it more space to do more work.
> We heard of a hardware (PCI CARDS) can be used with Asterisk that does the
> work. And we stumbled with Digium TC400B.
> Could be a newbie's question.. but does that serve our needs? As we have not
> pressured a server before up to 1400 extensions with 600 outbound SIP calls
> (customer's needs).
> The server in question is Core I7 16 GB ram and Raid 10 SAS drives.
> We need to know how many calls with G729 or G723 can this server handle? And
> as far as we can see this Digium card can be a cheaper solution If
> calculating the CPU cost plus the licenses for each channel.
> One more question.. can we add two of those cards to the server? Will it be
> efficient?

Hi Tarek,

I can not address your question about how many software transcoding
sessions your server can handle, but I can say that the TC400M works
fine with multiple cards in a single server.  The driver spreads the
load among installed cards.

Cheers,
Shaun

-- 
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Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk and Digium TC400B

[Bryant Zimmerman]

We can resell the Sangoma card. They have some higher license counts as 
well.
They are also offering a step up offering. If you buy at one level and need 
to move to the next.
They will offer you a trade back on the old card.

Bryant


 From: "Tim Nelson" 
Sent: Thursday, September 23, 2010 10:05 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Asterisk and Digium TC400B

- "Tarek Sawah"  wrote:
> Greetings,
> Because of the heavy load and the high expectations of an asterisk
> server
> offered as a solution integrated with our CRM software.. we were
> looking
> into other possibilities than software Licenses for G729 and G723
> codecs..
> to lower the pressure on the processor giving it more space to do more
> work.
> We heard of a hardware (PCI CARDS) can be used with Asterisk that does
> the
> work. And we stumbled with Digium TC400B.
> Could be a newbie's question.. but does that serve our needs? As we
> have not
> pressured a server before up to 1400 extensions with 600 outbound SIP
> calls
> (customer's needs).
> The server in question is Core I7 16 GB ram and Raid 10 SAS drives.
> We need to know how many calls with G729 or G723 can this server
> handle? And
> as far as we can see this Digium card can be a cheaper solution If
> calculating the CPU cost plus the licenses for each channel.
> One more question.. can we add two of those cards to the server? Will
> it be
> efficient?

Sangoma also has a transcoding card:

http://sangoma.com/products/hardware_products/transcoding.html

My understanding of both the Digium and Sangoma offerings is that you can 
use multiple cards in your system.

--Tim

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Re: [asterisk-users] realm: security issue

[Zeeshan Zakaria]

>From what you explained it seems to me that your mobile provider has blocked
your sip communication altogether. Have you tried changing IP address of
your asterisk server? If changing IP works, then probably your provider has
blocked you sip communication by IP only.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-23 7:22 AM, "bilal ghayyad"  wrote:

Hi All;

I have my friend that use his mobile (Nimbuz) to connect for the Asterisk
and his account was working fine. Suddenly it stop working (not able to
register).

>From my mobile (Nokia) I was able to register using my username and
password, so I tried to register using his (my friend) username and password
(that was using them from Nimbuz), it did not work. I come back trying to
register using my origin username and password (which was working fine just
before a while), it did not work. I removed my username and my friend
username from the Asterisk and then I created a new username and password
(different than all other) and I tried to register from my mobile, also it
did not work !!!

I start beleive that it is something related to detecting a hacking (maybe
Nimbuz does not use a good security), this caused the MAC to be considered
as hacked.

Please, can someone advise me how to resolve this problem? Where I can find
those MACs that need to be removed from block list? What can I do to get out
from this problem?

Any advise?
Regards
Bilal




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[asterisk-users] Sip from ip address

[Geraint Lee]

Is there a way to specify which IP address to originate calls from in a peer
on sip.conf?

I need to send calls from 10.1.3.10 which is a routed network through
openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk
box is the same box as the vpn bridge for the 10.1.3.0/24 network. I can't
set the host as 10.39.x.x as it is dynamic.

i can't change bindaddr since i need to be able to receive connections from
the external ip address as well as the internal address - unless there's a
way to specify 2 ip's to use?

For now i will use "friend" with a dynamic host instead of peer, but would
prefer to use peer without having to use username and passwords.

Cheers

Geraint
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Re: [asterisk-users] Record() Cmd and My SQL

[Govind, Mahesh (NSN - IN/Bangalore)]

Thanks .

I checked in voip info but I was not able to find a script which records
directly to database . All scripts were recording directly to files .

 

About #2 and  #3 can you please provide some more information .

 

Regards

Mahesh

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext Danny
Nicholas
Sent: Thursday, September 23, 2010 7:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Record() Cmd and My SQL

 

Just my .02 - 

#1 - voip-info.org is a good resource for finding this kind of script

#2 - I think you are asking for trouble doing a direct recording using
the PHP/MySQL combination; Personally I would do C/MySQL but the
exposure you face depends on the length of your recording(s).

#3 - since you are going to need to go the AGI route anyway, why not do
a MixMonitor/AGI combination so you get the speed/reliability of a disk
write-to-file?

 

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Re: [asterisk-users] Net2Phone SIP trunk problem

[Gopalakrishnan A.N]

Please give a try like the following,

Xlite (Configure Net2Phone A/c) ---> Cisco ASA Firewall ---> Internet cloud,

if the above works then there is no problem with your firewall, replace the
nat=yes and canreinvite=yes

otherwise you have to allow the ports 5060 (TCP), 5000 to 3 (UDP) in
your router for the Asterisk IP (192.168.0.10)

Try this...

On Thu, Sep 23, 2010 at 7:33 PM, Alejandro Cabrera Obed
wrote:

> Dear, I have this scenario:
>
> - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10
>
> - Behind a Cisco ASA firewall that connects to Internet
>
> - SIP trunk to Net2Phone with these parameters (nat=no):
>
> host=200.58.113.60
> username=DOLLY
> secret=123456
> port=5060
> type=peer
> dtmfmode=rfc2833
> disallow=all
> allow=alaw&ulaw
> nat=no
> canreinvite=no
> qualify=yes
>
> -Softphones Xlite
>
> The PBX can't register to Net2Phone, and no calls are made and this is the
> log:
>
>  -- Executing [...@macro-dialout-trunk:20] NoOp("SIP/9004-0008", "Dial
> failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20")
> in new stack
> -- Executing [...@macro-dialout-trunk:21] Goto("SIP/9004-0008",
> "s-CHANUNAVAIL,1") in new stack
> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
> -- Executing [s-chanunav...@macro-dialout-trunk:1]
> Set("SIP/9004-0008", "RC=20") in new stack
> -- Executing [s-chanunav...@macro-dialout-trunk:2]
> Goto("SIP/9004-0008", "20,1") in new stack
> -- Goto (macro-dialout-trunk,20,1)
> -- Executing [...@macro-dialout-trunk:1] Goto("SIP/9004-0008",
> "continue,1") in new stack
> -- Goto (macro-dialout-trunk,continue,1)
> -- Executing [conti...@macro-dialout-trunk:1]
> GotoIf("SIP/9004-0008", "1?noreport") in new stack
> -- Goto (macro-dialout-trunk,continue,3)
> -- Executing [conti...@macro-dialout-trunk:3]
> NoOp("SIP/9004-0008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE:
> 20 - failing through to other trunks") in new stack
> -- Executing [conti...@macro-dialout-trunk:4] Set("SIP/9004-0008",
> "CALLERID(number)=9004") in new stack
>
> What can be the problem please ???
>
> Thanks a lot
>
> Alejandro
>
>
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Gopalakrishnan A.N,
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[asterisk-users] Can't turn debug on in a 1.2 box

[khalid touati]

Hi Guys,
i could turn debug on in a asterisk 1.6 box (by enabling debug in
logger.conf and core set debug to > 0), but my issue is i cannot enable
debugging in a 1.2 box by doing the same 2 steps, also this is a production
server so i can't restart with debug enabled, do you guys know how i can
turn debug on or just know why it's not getting enabled?
Thanks a lot for your help!
-- 
Abdullah
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Re: [asterisk-users] Asterisk and Digium TC400B

[Tim Nelson]

- "Tarek Sawah"  wrote:
> Greetings,
> Because of the heavy load and the high expectations of an asterisk
> server
> offered as a solution integrated with our CRM software.. we were
> looking
> into other possibilities than software Licenses for G729 and G723
> codecs..
> to lower the pressure on the processor giving it more space to do more
> work.
> We heard of a hardware (PCI CARDS) can be used with Asterisk that does
> the
> work. And we stumbled with Digium TC400B.
> Could be a newbie's question.. but does that serve our needs? As we
> have not
> pressured a server before up to 1400 extensions with 600 outbound SIP
> calls
> (customer's needs).
> The server in question is Core I7 16 GB ram and Raid 10 SAS drives.
> We need to know how many calls with G729 or G723 can this server
> handle? And
> as far as we can see this Digium card can be a cheaper solution If
> calculating the CPU cost plus the licenses for each channel.
> One more question.. can we add two of those cards to the server? Will
> it be
> efficient?

Sangoma also has a transcoding card:

http://sangoma.com/products/hardware_products/transcoding.html

My understanding of both the Digium and Sangoma offerings is that you can use 
multiple cards in your system.

--Tim

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Re: [asterisk-users] Asterisk T38

[Matt Watson]

On Wed, Sep 22, 2010 at 10:46 AM, Adam Moffett  wrote:

> That's probably what I'm going to have to do.  Thanks.
>
> > I suppose that merely removing ATA and asterisk from the middle, and
> > plugging a pots line into a fax machine is out of the question.
> >
> >
>
>
>
If you are running 1.6, you can try this -
https://issues.asterisk.org/view.php?id=13405  I have this implemented on
our production system but I personally get hit and miss results with it.
 Sometimes it doesn;t seem to detect its a fax call.   The fax typically
goes through fine over g.711u though as long as you have good network links.
 My setup previously included a couple of lan-extensions over DSL and it was
pretty unreliable on them if there was other network traffic going on even
with QoS on the edge routers heavily favoring the phone traffic.

If you are running 1.4 you
can try this - http://www.zoiper.com/foip/  I used this back before
zoiper acquired the company that owned it and it worked absolutely perfect.
 I haven;t tried it since its been open sourced and had 3rd party people
patch it current versions of asterisk though.
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[asterisk-users] Net2Phone SIP trunk problem

[Alejandro Cabrera Obed]

Dear, I have this scenario:

- PBX Asterisk 1.6.2.10 with private IP 192.168.0.10

- Behind a Cisco ASA firewall that connects to Internet

- SIP trunk to Net2Phone with these parameters (nat=no):

host=200.58.113.60
username=DOLLY
secret=123456
port=5060
type=peer
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw
nat=no
canreinvite=no
qualify=yes

-Softphones Xlite

The PBX can't register to Net2Phone, and no calls are made and this is the
log:

-- Executing [...@macro-dialout-trunk:20] NoOp("SIP/9004-0008", "Dial
failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20")
in new stack
-- Executing [...@macro-dialout-trunk:21] Goto("SIP/9004-0008",
"s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-chanunav...@macro-dialout-trunk:1]
Set("SIP/9004-0008", "RC=20") in new stack
-- Executing [s-chanunav...@macro-dialout-trunk:2]
Goto("SIP/9004-0008", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [...@macro-dialout-trunk:1] Goto("SIP/9004-0008",
"continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [conti...@macro-dialout-trunk:1]
GotoIf("SIP/9004-0008", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [conti...@macro-dialout-trunk:3] NoOp("SIP/9004-0008",
"TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to
other trunks") in new stack
-- Executing [conti...@macro-dialout-trunk:4] Set("SIP/9004-0008",
"CALLERID(number)=9004") in new stack

What can be the problem please ???

Thanks a lot

Alejandro
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Re: [asterisk-users] Record() Cmd and My SQL

[Danny Nicholas]

Just my .02 - 

#1 - voip-info.org is a good resource for finding this kind of script

#2 - I think you are asking for trouble doing a direct recording using the
PHP/MySQL combination; Personally I would do C/MySQL but the exposure you
face depends on the length of your recording(s).

#3 - since you are going to need to go the AGI route anyway, why not do a
MixMonitor/AGI combination so you get the speed/reliability of a disk
write-to-file?

 

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Re: [asterisk-users] Record() Cmd and My SQL

[Gopalakrishnan A.N]

sorry I dont have samples, but you can find lots of php agi scripts by
googling..

On Thu, Sep 23, 2010 at 7:07 PM, Govind, Mahesh (NSN - IN/Bangalore) <
mahesh.gov...@nsn.com> wrote:

>  Thanks ,
>
> Do you have some sample for that .
>
> Regards
>
> Mahesh
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *ext Gopalakrishnan
> A.N
> *Sent:* Thursday, September 23, 2010 5:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Record() Cmd and My SQL
>
>
>
> I hope it cant be done using dialplan but it can be done through AGI
> scripting...you can use your favorite programming language PHP / Perl and
> try to record the call with record application and point to the database
> instead of a folder path.
>
>
>
> Asterisk Users your feedback also welcome on this..
>
> On Thu, Sep 23, 2010 at 11:51 AM, Govind, Mahesh (NSN - IN/Bangalore) <
> mahesh.gov...@nsn.com> wrote:
>
> HI ,
>
> Is there Any way is there so that I can store my recordings directly to a
> database rather storing the same to a file .
>
> Thanks in advance .
>
> Regards
>
> Mahesh
>
>
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> _
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>
>
>
>
> --
> Thank you  with regards,
> Gopalakrishnan A.N,
>
>
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Re: [asterisk-users] Record() Cmd and My SQL

[Govind, Mahesh (NSN - IN/Bangalore)]

Thanks ,

Do you have some sample for that .

Regards

Mahesh

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext
Gopalakrishnan A.N
Sent: Thursday, September 23, 2010 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd and My SQL

 

I hope it cant be done using dialplan but it can be done through AGI
scripting...you can use your favorite programming language PHP / Perl
and try to record the call with record application and point to the
database instead of a folder path.

 

Asterisk Users your feedback also welcome on this..

On Thu, Sep 23, 2010 at 11:51 AM, Govind, Mahesh (NSN - IN/Bangalore)
 wrote:

HI , 

Is there Any way is there so that I can store my recordings directly to
a database rather storing the same to a file .

Thanks in advance .

Regards

Mahesh


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Gopalakrishnan A.N,



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Re: [asterisk-users] func SHARED, how to use?

[Philipp von Klitzing]

Hi!

> There are 2 things I can't understand
> - 1. how can I know channel name?

${CHANNEL}

> 2. where should I call this SHARED function? before Dial, after Dial?

Either In the macro that you specify using the M option of Dial() or in 
the h extension. You will, however, have trouble treating the 2nd call 
leg with the h extension, and the M option only "fires" at the beginning 
of the call/bridge.

> and this is what I tried in h:
> exten => h,1,Set(CDR(userfield)=${CHANNEL(rtpqos,audio,local_count)})
>
> This can looks different, this is just a try :-)
> But Channel is DAHDI, not SIP, so I want to use SHARED.

Are you on Asterisk 1.4 or 1.6 or on 1.8? The above will not work 
with 1.4. 

Since I have not done this with 1.6 or 1.8: See if you can get the RTCP 
data without using CHANNEL(), and instead use the individual "xxxBRDIGED" 
RTCP channel variables as illustrated on the Wiki. Your SIP channel is 
the 2nd channel (= the bridged one).  

If this fails: See if AMI can help.

Philipp


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Re: [asterisk-users] func SHARED, how to use?

[Dmitry Melekhov]

23.09.2010 16:06, Philipp von Klitzing пишет:
> Hi!
>
>
>>> And the third hit in my google result is this:
>>>
>>> http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
>>>
>>> Since I mentioned in my previous message that you will find the answer
>>> in the archive of this list you could have found that even without
>>> google. gmane.org for example has a nice web UI for reading this list.
>>>
> >
>
>> I'm sorry, but this is absolutely the same thing I see on voip-info.org.
>>  
> Indeed, I added that info to the Wiki yesterday to make it easier for you
> in case you would not succed with Google.
>
>

Well, this doesn't make things simpler for me :-(

> Since you are not describing a concrete problem in detail (and illustrate
> that with some code) it is difficult to help you. For example you still
> do not tell us if the Zap/DAHDI channel is your first or second call leg,
> and you do not tell us which data you would like to access, and how it is
> failing in your current implementation.
>

DAHDI is channel from which I call Dial, so it is only accessible in h.
Otherwise I'll not ask this question :-)
> - SIP/123 might be your Zap/DAHDI channel, which you could pass onto the
> second call leg either by a channel variable prefixed with an underscore
> (like _MYVAR), or by adding it as a Macro argument.
>
>
There are 2 things I can't understand-
1. how can I know channel name? imho, this will be far easier to have 
something to name bridged channel...
But I guess it is very simple , I just don't know how...
2. where should I call this SHARED function? before Dial, after Dial?

Here is my calling code:
exten =>_10,n,Dial(SIP/${EXTEN:2...@peer,,g)
exten =>_10,n,Hangup()

...
and this is what I tried in h:

exten => h,1,Set(CDR(userfield)=${CHANNEL(rtpqos,audio,local_count)})

This can looks different, this is just a try :-)
But Channel is DAHDI, not SIP, so I want to use SHARED.

Thank you!



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Re: [asterisk-users] Record() Cmd and My SQL

[Gopalakrishnan A.N]

I hope it cant be done using dialplan but it can be done through AGI
scripting...you can use your favorite programming language PHP / Perl and
try to record the call with record application and point to the database
instead of a folder path.

Asterisk Users your feedback also welcome on this..

On Thu, Sep 23, 2010 at 11:51 AM, Govind, Mahesh (NSN - IN/Bangalore) <
mahesh.gov...@nsn.com> wrote:

>  HI ,
>
> Is there Any way is there so that I can store my recordings directly to a
> database rather storing the same to a file .
>
> Thanks in advance .
>
> Regards
>
> Mahesh
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Gopalakrishnan A.N,
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Re: [asterisk-users] func SHARED, how to use?

[Philipp von Klitzing]

Hi!

> > And the third hit in my google result is this:
> >
> > http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
> >
> > Since I mentioned in my previous message that you will find the answer
> > in the archive of this list you could have found that even without
> > google. gmane.org for example has a nice web UI for reading this list.
   > 
> I'm sorry, but this is absolutely the same thing I see on voip-info.org.

Indeed, I added that info to the Wiki yesterday to make it easier for you 
in case you would not succed with Google.

> And, I'm too stupid to understand how to use it in dial plan, especially
> for RTCP statistics. :-( May be this is very-very simple, so nobody
> understand what I want, if it is absolutely clear... But,could someone
> provide me example of how to use SHARED with RTCP?

Since you are not describing a concrete problem in detail (and illustrate 
that with some code) it is difficult to help you. For example you still 
do not tell us if the Zap/DAHDI channel is your first or second call leg, 
and you do not tell us which data you would like to access, and how it is 
failing in your current implementation.

Anyway, usage of SHARED _is_ very simple: In the Macro that is called 
with the M option of the Dial() statement you add a one-line statement of 
the type

  Set(SHARED(foo,SIP/123)=456)

- foo is the variable name that you choose
- 456 is the data you are interested in (e.g. the codec chosen)
- SIP/123 might be your Zap/DAHDI channel, which you could pass onto the 
second call leg either by a channel variable prefixed with an underscore 
(like _MYVAR), or by adding it as a Macro argument.

As a result you will be able to use 

   ${SHARED(foo,SIP/123)} 

in the h extension of your first call leg.

Note: RTCP data are only available for SIP channels, not for Zap/DADHI 
channels or other channel technologies. Information about the negotiated 
codec is not part of the RTCP data.

Philipp


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[asterisk-users] Asterisk and Digium TC400B

[Tarek Sawah]

Greetings,
Because of the heavy load and the high expectations of an asterisk server
offered as a solution integrated with our CRM software.. we were looking
into other possibilities than software Licenses for G729 and G723 codecs..
to lower the pressure on the processor giving it more space to do more work.
We heard of a hardware (PCI CARDS) can be used with Asterisk that does the
work. And we stumbled with Digium TC400B.
Could be a newbie's question.. but does that serve our needs? As we have not
pressured a server before up to 1400 extensions with 600 outbound SIP calls
(customer's needs).
The server in question is Core I7 16 GB ram and Raid 10 SAS drives.
We need to know how many calls with G729 or G723 can this server handle? And
as far as we can see this Digium card can be a cheaper solution If
calculating the CPU cost plus the licenses for each channel.
One more question.. can we add two of those cards to the server? Will it be
efficient?
Regards
Tarek Sawah



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[asterisk-users] realm: security issue

[bilal ghayyad]

Hi All;

I have my friend that use his mobile (Nimbuz) to connect for the Asterisk and 
his account was working fine. Suddenly it stop working (not able to register).

>From my mobile (Nokia) I was able to register using my username and password, 
>so I tried to register using his (my friend) username and password (that was 
>using them from Nimbuz), it did not work. I come back trying to register using 
>my origin username and password (which was working fine just before a while), 
>it did not work. I removed my username and my friend username from the 
>Asterisk and then I created a new username and password (different than all 
>other) and I tried to register from my mobile, also it did not work !!!

I start beleive that it is something related to detecting a hacking (maybe 
Nimbuz does not use a good security), this caused the MAC to be considered as 
hacked.

Please, can someone advise me how to resolve this problem? Where I can find 
those MACs that need to be removed from block list? What can I do to get out 
from this problem?

Any advise?
Regards
Bilal


  

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Re: [asterisk-users] CDR display in minute

[Mickael MONSIEUR]

http://forums.cacti.net/viewtopic.php?p=111317

Thank you.

2010/9/23 Faisal Hanif 

>  use CACTI
>
> On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:
>
> Hello,
> I want to graphically display the number of calls per minute to an
> extension.
>
> The programs I have found it possible to do so but the average is done on
> time or day ...
>
> I use Mysql CDR
>
> Thank you,
> Mickael
>
>
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Re: [asterisk-users] CDR display in minute

[Faisal Hanif]

 use CACTI

On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:

Hello,
I want to graphically display the number of calls per minute to an 
extension.


The programs I have found it possible to do so but the average is done 
on time or day ...


I use Mysql CDR

Thank you,
Mickael
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[asterisk-users] CDR display in minute

[Mickael MONSIEUR]

Hello,
I want to graphically display the number of calls per minute to an
extension.

The programs I have found it possible to do so but the average is done on
time or day ...

I use Mysql CDR

Thank you,
Mickael
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Re: [asterisk-users] Recording maximum time and stop on silence

[David Cunningham]

Danny, thank you!


On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas  wrote:

>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
> *Sent:* Wednesday, September 22, 2010 4:28 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Recording maximum time and stop on silence
>
>
>
> All,
>
> Two questions:
>
> 1. Is there a limit on how long a call can be recorded for? For example is
> 4 hours a problem?
>
> 2. Can recording be stopped after a configured period of silence?
>
> Thanks in advance,
>
> --
> David Cunningham, Voisonics
> http://voisonics.com/
> US toll-free: +1 888 842 2720
> UK: +44 (0) 20 3298 1642
> Australia: +61 (0) 2 9037 2180
>
> AFAIK, #1 is limited only by available disk space, #2 is yes, but you may
> have to tweak some settings to “get it right”
>
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-- 
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http://voisonics.com/
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UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
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Re: [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib

[IMS]

Hi,

Excuse me if I'm late to reply but my first response has been blocked by the
moderator (message too big)
So I've created an account on rapidshare to share my config.log and
menuselect/config.log
Hope it will help.

The link : http://rapidshare.com/users/Z8SX25

Thanks for any help !

Sebastien


On Wed, Sep 22, 2010 at 9:21 AM, IMS  wrote:
> Do you have any ideas of the problem ? config.log don't give me more
> explanations.
>
Attach your config.log so we can see what is going on.

--
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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