Re: [asterisk-users] Capture Media IP in CDR

[Warren Selby]

On Fri, Oct 11, 2013 at 9:05 PM, CDR  wrote:

> I am not proxying the media, but never the less I am forced to store
> the source media IP in my CDR, for regulatory reasons. Asterisk gets
> that information when the reinvite comes, but how do I store it?
> If I don't figure this out my next email will be from Federal Prison.
> Kindly help me stay away from those guys. Eventually we all need to
> save that information or we shall not be able to stay in business.
>
>

You can add custom fields to your CDR records using
Set(CDR(customfieldname)=foobar).  I don't know the name of the variable
you want that specifically contains the source media IP, but I imagine you
can pull it with the SIP_HEADER function, or possibly the CHANNEL(recvip)
function.


--
Thanks,
Warren Selby, dCAP
http://www.SelbyTech.com 
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[asterisk-users] Capture Media IP in CDR

[CDR]

I am not proxying the media, but never the less I am forced to store
the source media IP in my CDR, for regulatory reasons. Asterisk gets
that information when the reinvite comes, but how do I store it?
If I don't figure this out my next email will be from Federal Prison.
Kindly help me stay away from those guys. Eventually we all need to
save that information or we shall not be able to stay in business.

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[asterisk-users] DTMF detection problem with analog card

[mohsen feyzzadeh]

Hi all.
I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port).
When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not 
receive DTMF from caller while the voice is playing. But if user waits to the 
end of playing voice, there is no problem.

I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4.
Could you please help me?
Here is my configs:

system.conf:

fxsks=1
fxsks=2
loadzone    = nl
defaultzone    = nl

chan_dahdi.conf:
--
[channels]
;===
;General options
;===
usecallerid = yes
hidecallerid = no
busydetect=yes
busycount=3

;===
;FXO Modules
;===
group = 1
signalling = fxs_ks
context = my-context
channel = 1,2-- 
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Re: [asterisk-users] Failed to authenticate user 1000; tag=03f82bb9

[Tiago Geada]

Hi,

Seems a great workaround from Gareth Blades. Thanks I will try it.

Any way to make asterisk log a line in /var/log/messages ?


On 10 October 2013 19:44, Michelle Dupuis  wrote:

>  Gareth:
>
> Did you check if your message (or security) log recorded anything during
> these attempts?  If so, can you post the content of the logs during this
> attack?
>
> M
>  --
> *From:* asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad [
> asghar...@gmail.com]
> *Sent:* Tuesday, October 01, 2013 11:53 AM
> *To:* Asterisk Users List
> *Subject:* Re: [asterisk-users] Failed to authenticate user
> 1000; tag=03f82bb9
>
>   Hi,
> Bad boys trying to guess a valid username.
> in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st
> invite.
>
>
> On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades <
> mailinglist+aster...@dns99.co.uk> wrote:
>
>> On 01/10/13 15:44, gincantalupo wrote:
>>
>> On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo <
>> gincantal...@fgasoftware.com> wrote:
>>
>>> Hi,
>>>
>>> I get a lot of these messages on my Asterisk CLI:
>>>
>>> "Failed to authenticate user 1000
>>> ;tag=03f82bb9"
>>>
>>> as if my PBX machine is trying to authenticate to itself. It seems
>>> someone is attacking my asterisk PBX.
>>>
>>> Is there a way to fix this problem?
>>
>>
>> in sip.conf I have guest connections permitted and have them going to the
>> default context which contains :-
>>
>> [default]
>> ; all unauthenticated connection attempts from the internet come in here.
>> exten => _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
>> ${SIP_HEADER(Contact)})
>> exten => _[+*#0-9].,n,Congestion
>>
>> Then in fail2ban I have it match the following :-
>>
>> failregex = Registration from .* failed for \'\' - Wrong password
>> Unauthenticated call attempt .*\@\:
>>
>>
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>
>
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Re: [asterisk-users] asterisk 11.6 nat problem

[Matthew J. Roth]

Jeremy Kister wrote:
> 
> using asterisk 11.6.0-rc1 i just converted my "nat=yes" to 
> "nat=auto_force_rport,auto_comedia"
> 
> I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
> 
> a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). 
>   A 'sip reload' always fixes the problem.
> 
> i left 'sip set debug peer vgw1' on the console.  but i dont see what's 
> causing the issue..
> 
> http://kister.net/tmp/ast-sip.conf
> http://kister.net/tmp/ast-console.txt
> 
> can anyone spot the issue?


Jeremy,

It looks like at some point Asterisk decides that vgw1's SIP port is
no longer 5060.  This may have to do with the NAT settings for that
device:

 Before 'sip reload'  |  After 'sip reload'
--|-
* Name   : vgw1   | * Name   : vgw1
...   | ...
Force rport  : Auto (Yes) | Force rport  : Auto (No)
Symmetric RTP: Auto (Yes) | Symmetric RTP: Auto (No)
...   | ...
Addr->IP : 10.9.1.9:59934 | Addr->IP : 10.9.1.9:5060
...   | ...
Status   : UNREACHABLE| Status   : OK (19 ms)

Since the device is on the same subnet as your Asterisk server, you
could try setting 'nat=no' for the vgw1 peer.  That may not be a good
long-term solution because of its security implications¹, but it
could help determine if the NAT settings are the cause of the problem
and serve as a stopgap until you figure out why the port is changing.

Alternatively, you could try setting 'port=5060' for the vgw1 peer,
but that's the default so it may still get changed.

¹ From sip.conf.sample:

  IT IS IMPORTANT TO NOTE that if the nat setting in the general
  section differs from the nat setting in a peer definition, then the
  peer username will be discoverable by outside parties as Asterisk
  will respond to different ports for defined and undefined peers. For
  this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
  GENERAL SECTION. Specifically, if nat=force_rport in one section and
  nat=no in the other, then valid peers with settings differing from
  those in the general section will be discoverable.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] chan_sip.c:9602 copy_header: No field 'CSeq' present to copy

[Todd R .]

Just put a new phone in place with the latest firmware from Cisco. This is the 
first SPA501G we have with this firmware.
In the Asterisk CLI we are now seeing the error message below about once every 
second. When we unplug the phone, the messages quit.










NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy


Thanks in advance for any assistance on this.   
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Re: [asterisk-users] GSM to SIP Adapter

[Tarek Sawah]

Thank you for the reply, actually we are looking for something like the 
followinghttp://www.ebay.com/itm/GSM1SIP-GSM-over-IP-GoIP-SIP-Quad-Bands-voip-gateway-Quad-band-1XGSM-GoIP-VoIP-/181075100268how
 ever our requirement are a bit wire like SMS in addition to Call capability.



Tarek Sawah




> From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Fri, 11 Oct 2013 15:33:36 +0100
> Subject: Re: [asterisk-users] GSM to SIP Adapter
> 
> On Friday 11 October 2013, Tarek Sawah wrote:
> > Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel
> > (one SIM card). any suggestions?
> > 
> > Tarek Sawah
> 
> We've been using OpenVox G400P cards  (PCI; there is also a G400E, which is 
> PCI express for newer motherboards).  Sends and receives text messages, and 
> makes and answers phone calls.  Accepts up to four RF modules, each of which 
> accepts one SIM card.
> 
> If you only need text message functionality  (not voice calls),  then almost 
> any old mobile phone with a USB or RS232 cable can be used as a GSM modem -- 
> and you probably have one lying in a drawer.
> 
> -- 
> AJS
> 
> Answers come *after* questions.
> 
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Re: [asterisk-users] GSM to SIP Adapter

[A J Stiles]

On Friday 11 October 2013, Tarek Sawah wrote:
> Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel
> (one SIM card). any suggestions?
> 
> Tarek Sawah

We've been using OpenVox G400P cards  (PCI; there is also a G400E, which is 
PCI express for newer motherboards).  Sends and receives text messages, and 
makes and answers phone calls.  Accepts up to four RF modules, each of which 
accepts one SIM card.

If you only need text message functionality  (not voice calls),  then almost 
any old mobile phone with a USB or RS232 cable can be used as a GSM modem -- 
and you probably have one lying in a drawer.

-- 
AJS

Answers come *after* questions.

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[asterisk-users] GSM to SIP Adapter

[Tarek Sawah]

Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one 
SIM card). any suggestions?

Tarek Sawah


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Re: [asterisk-users] Read Telnet Packet

[A J Stiles]

On Friday 11 October 2013, akhilesh chand wrote:
> Dear All,
> 
> I want to read telnet packet continuously whenever a new call is originated
> and store into a variable after that pass into window server. I have
> written a Perl script to read telnet packet but problem is that whenever I
> executed Perl script then got a telnet packet( mean Only when i execute
> Perl script) here I want to put scheduler,event or other technique whenever
> a new call will come Perl script automatically run.

This is really a Perl question, not an Asterisk question, and you might have 
more joy asking on a Perl list or forum.

Anyway, there are two ways to accomplish what you want.  The easy way is to 
have inetd start your Perl script whenever a packet is sent to a particular 
port; when the script starts, its STDIN and STDOUT will already be connected 
to the port.  The other way is to have your Perl script run as a daemon; then 
fork off a clone of itself to deal with requests as they come in.  After a 
fork, the child process will inherit a copy of the socket object; and this 
will persist even after you undef it in the parent process.

If you want to invoke a Perl script from within the dialplan, you can either 
use a full-on AGI script  (which will even allow you to pass variables back 
and forth and do other funky stuff)  or use System() for a "quick and dirty" 
call to a script that doesn't need to send anything back.

-- 
AJS

Answers come *after* questions.

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