Re: [asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted

[Giorgio Incantalupo]

Hi Jonas,

I get this error when I incorrectly set my PBX gateway AND I have a sip 
peer trying to register outside (i.e.: a sip provider).
Are you sure about your sip.conf?

Giorgio Incantalupo


Jonas Kellens wrote:
> Hello,
>
> my Asterisk CLI is flooded with the following message :
>
> [Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
> of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
> not permitted
> [Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
> of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
> not permitted
> [Jun 25 21:25:05] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
> of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
> not permitted
> [Jun 25 21:25:09] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
> of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
> not permitted
> [Jun 25 21:25:13] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
> of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
> not permitted
>
>
> I have no idea where this IP comes from, there is no SIP peer or user 
> with this IP-address.
>
> What can I do to get ride of this message that is constantly flooding 
> my CLI ?!
>
>
> Reloading or restarting my Asterisk does not help !
>
>
> Kind regards,
>
> Jonas.


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[asterisk-users] RE How to break pri DID to multiple SIP Trunks

[Samantha]

Hey Guys

 

I have an indial range of 6128[01234]X  being trunked sip to
xxx.yyy.189.65

 

Now I want to break this down into 61280x going to xxx.yyy.188.145 and
61284x going to xxx.yyy.189.199
reminder being used for fax->email  etc etc etc

 

I have created the outbound routes and sip trunks

I can see that all the sip trunks are up 
I can see the outbound routes are there and also in trunks

 

But it isn't working

 

The call gets answered by the first point xxx.yyy.189.69 and you get an rva
of the number you called is not in service

 

 

Regards

 

Samantha

 

 

 

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[asterisk-users] Long shot... Order Logix

[Matt Desbiens]

Has anyone ever integrated the software from order logix into their system?
This is primarily an API driven, pulled from a SQL database and stored for a
client to access... Order Logix deals primarily with Call Centers, it pulls
the information from the SQL database, and will allow access for the client
to pull the recording and all associated call information...  I know its a
long shot and everything should be in SQL to be pulled from the DB and
posted, but I want to know what I'm getting into before I dive in...

-- Matt
//* EOF *//
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Re: [asterisk-users] Codec negotiation

[Ryan Wagoner]

On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing
 wrote:
> Hi!
>
>> Because the codec is already chosen before the call is made, and you
>> told it that g722 is permitted.
>>
>> There are all sorts of discussions in play about codec negotiation,
>> but at this point in time, if you want different behaviour you'll need to
>> go and code it yourself
>
> Look at the list archive - there is a codec negotiation patch around:
>
> http://lists.digium.com/pipermail/asterisk-users/2010-
> February/244835.html
>
> The OP might also want to consider to use different lines to the same
> PBX, one for normal narrowband, and another one for g722.
>
> Philipp
>
>
> --

Thanks! I'm going to try setting the _SIP_CODEC variable for outbound
calls to force ulaw. This should solve the issue. Having two lines
would work but I can't sell this to a customer. There has got to be a
better way to have Asterisk handle this. With Asterisk in the middle
of the RTP stream it knows what both parties support. If it turns out
Asterisk is transcoding it could check for a common codec and
renegotiate one endpoint.

Ryan

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[asterisk-users] Anyone can share their config file for Cisco phone please?

[bruce bruce]

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been
flashed with SIP firmware but the config file doesn't seem to work maybe I
am missing something in it.

I appreciate it if you can share your working sample config file with me.

Thanks
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Re: [asterisk-users] Codec negotiation

[Philipp von Klitzing]

Hi!

> Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??

Most probably - who on this list would you like to test it for you? ;->

Philipp


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Re: [asterisk-users] Codec negotiation

[Philipp von Klitzing]

Hi!

> Because the codec is already chosen before the call is made, and you
> told it that g722 is permitted.
> 
> There are all sorts of discussions in play about codec negotiation,
> but at this point in time, if you want different behaviour you'll need to
> go and code it yourself

Look at the list archive - there is a codec negotiation patch around:

http://lists.digium.com/pipermail/asterisk-users/2010-
February/244835.html

The OP might also want to consider to use different lines to the same 
PBX, one for normal narrowband, and another one for g722.

Philipp


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Re: [asterisk-users] Dial options not working

[Anahi Ludueña]

Thanks, but I don't have any *dahdi*.conf file here... (I check in 
/etc/asterisk)




Anahi Ludueña
 



From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 29 Jun 2010 16:54:01 -0500
Subject: Re: [asterisk-users] Dial options not working



















Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi
files this lives in).  Sounds like your DAHDI doesn’t like DTMF input.

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Tuesday, June 29, 2010 4:51
PM

To: asterisk-users@lists.digium.com

Subject: [asterisk-users] Dial
options not working



 

Hi, I have an extension which has
the follow me option activated. The followme option should go to a IVR if no
answer...

The problem that I have is that everything works when I'm calling it from my
extension, but if I use any landline phone or a cell phone, I'm unable to enter
any options. When I press one option, it seems I do nothing...

Please, could you help me?

Thanks,















Anahi
Ludueña

 















Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo
ya!

  
_
Citas sin compromiso por Internet Te damos las claves para encontrar pareja en 
la red
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Re: [asterisk-users] Dial options not working

[Danny Nicholas]

Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files
this lives in).  Sounds like your DAHDI doesn’t like DTMF input.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Tuesday, June 29, 2010 4:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial options not working

 

Hi, I have an extension which has the follow me option activated. The
followme option should go to a IVR if no answer...
The problem that I have is that everything works when I'm calling it from my
extension, but if I use any landline phone or a cell phone, I'm unable to
enter any options. When I press one option, it seems I do nothing...
Please, could you help me?
Thanks,




  _  

Anahi Ludueña

 





  _  

Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo
  ya!

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Matt Darnell]

Thank you Andrew,

I will check it out.  We are currently running 1.4.

-Matt

On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham  wrote:
> Remote Party ID in trunk, it works  There are hacks for other versions.
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
> * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
> * Learn more about Linux http://en.wikipedia.org/wiki/Linux
> * Learn more about Tux http://en.wikipedia.org/wiki/Tux
>
>
>
> On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell  wrote:
>> Is is possible with a Polycom phone to update the LCD with the
>> callee's name after dialing them?
>>
>> When you dial ext 103 now, it says 'To:103'...would be nice if could
>> have 'To:Dan Marino'
>>
>> This is the case even when you have a contact for ext 103.
>>
>> None of the phones I have ever tested do this, Polycom, Linksys,
>> Cisco, Grandstream, Yealink, etc.
>>
>> -Matt
>>
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[asterisk-users] Dial options not working

[Anahi Ludueña]

Hi, I have an extension which has the follow me option activated. The followme 
option should go to a IVR if no answer...
The problem that I have is that everything works when I'm calling it from my 
extension, but if I use any landline phone or a cell phone, I'm unable to enter 
any options. When I press one option, it seems I do nothing...
Please, could you help me?
Thanks,






Anahi Ludueña
 

  
_
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Re: [asterisk-users] Strange Asterisk/SIP call forwarding behavior

[Paul Belanger]

On Tue, Jun 29, 2010 at 2:36 PM, Myles Wakeham  wrote:
> What I'm looking for is some sort of advice on 'best practice' to handle
> call routing to my cell phone vs. keeping it on my LAN to my desktop
> phone.  As I mentioned, the desktop phone works flawlessly and I'm
> trying to get the same results on my cell phone so I don't lose any
> calls in the process.
>
*CLI> core show application FollowMe

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Re: [asterisk-users] SIP Delay with remote stations?

[Paul Belanger]

On Tue, Jun 29, 2010 at 10:06 AM, William Stillwell (Lists)
 wrote:
> Any idea what could be causing this?
>
Yes, network delay, packet loss, the Internet.  Implement QoS and
bandwidth monitoring.

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Re: [asterisk-users] Carrier needs more call examples

[Peder]

It depends on the issue.  If you have a carrier that has say 5-10 different
routes, they may want to confirm that the issue occurs on the same route
every time, or see if it is hitting the same box on their end.
Theoretically they could gather all the info on their end given the
caller/called numbers and times, but it is easier to make you get it.  I
have one carrier I deal with that pretty much won't accept a ticket without
a packet capture attached to it. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Tuesday, June 29, 2010 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Carrier needs more call examples

Ok list users, this is a question born out of curiosity, but if I'm 
having an intermittent problem and the carrier wants some examples of 
calls where the problem happened, what can they actually do with that 
information?

I guess my implementation is relatively simple here and all I've got to 
look at is CDR's and the asterisk log.  Does my carrier have 
tremendously more information they can look at?

I ask because I gave them 2 or 3 examples and they want more, and I 
don't know what difference it makes whether they have 3 examples to look 
at versus 300.


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[asterisk-users] Can't call my extension

[Nicholas Hart]

Hi,

I managed to get a remote extension to work through a router which can now
call all the other local extensions in asterisk.  For some reason, nobody
can call me back.  They get failed upon trying.  Keep thinking there must be
some caller group to which I need be added.  Or perhaps I need to add the IP
address of this phone to the sip.conf file?  Please let me know.  Thanks.

Nick
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Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs

[Paul Belanger]

On Tue, Jun 29, 2010 at 12:51 PM, Kenny Watson
 wrote:
> Is it simply a case of converting the prompts into other codecs and asterisk
> will pick these up?
>
Yes, install both g729 and ulaw/alaw prompts to avoid trans-coding altogether.

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[asterisk-users] Carrier needs more call examples

[Adam Moffett]

Ok list users, this is a question born out of curiosity, but if I'm 
having an intermittent problem and the carrier wants some examples of 
calls where the problem happened, what can they actually do with that 
information?

I guess my implementation is relatively simple here and all I've got to 
look at is CDR's and the asterisk log.  Does my carrier have 
tremendously more information they can look at?

I ask because I gave them 2 or 3 examples and they want more, and I 
don't know what difference it makes whether they have 3 examples to look 
at versus 300.


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Re: [asterisk-users] Codec negotiation

[Jonas Kellens]

Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??

I have reported a codec-issue, but there is no solution. Will this patch 
also answer my question ??

https://issues.asterisk.org/view.php?id=17020


Jonas.


On 06/29/2010 09:42 PM, Mindaugas Kezys wrote:

Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch

Regards,
Mindaugas Kezys

Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com


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[asterisk-users] Asterisk 1.6 (and 1.4) DTMF problems using RFC2833

[Mark Deneen]

We are experiencing intermittent DTMF problems here, with the following
setup:

ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF).

I am running Ubuntu server 10.04, but Asterisk is compiled by us and not
installed from the software repository.  Essentially, DTMF works for some
time, but at some point it simply stops and the point at which it stops
appears to be random.

Using RTP debug, I can verify that the RFC2833 DTMF is being delivered in
the RTP stream, and Asterisk knows of it.  Independently, wireshark confirms
the same.  I can't easily remove the PIX, but as the RTP is showing the DTMF
I do not believe the firewall is an issue.

Our ITSP is registered as a SIP provider, and we can receive calls just
fine.  I've attached a file containing portions of the asterisk log, the
wireshark log and the dialplan.

Has anyone else run into this situation?

Best Regards,
Mark Deneen
[Jun 29 15:31:44] DTMF[26287] channel.c: DTMF begin '*' received on SIP/10.200.10.5-001e
[Jun 29 15:31:44] DTMF[26287] channel.c: DTMF begin ignored '*' on SIP/10.200.10.5-001e
[Jun 29 15:31:44] DTMF[26287] channel.c: DTMF end '*' received on SIP/10.200.10.5-001e, duration 100 ms
[Jun 29 15:31:44] DTMF[26287] channel.c: DTMF end passthrough '*' on SIP/10.200.10.5-001e

[Jun 29 15:32:18] VERBOSE[26287] pbx.c: -- Timeout on SIP/10.200.10.5-001e, continuing...

(enable RTP debug)

Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017379, ts 5727107, len 04, mark 1, event 000b, end 0, duration 0)
[Jun 29 15:37:30] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:30] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050557, ts 5669136, len 20)
[Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017380, ts 5727107, len 04)
[Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017380, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00160)
[Jun 29 15:37:30] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050558, ts 5669296, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017381, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017381, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00320)
[Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050559, ts 5669456, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017382, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017382, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00480)
[Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050560, ts 5669616, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017383, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017383, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00640)
[Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050561, ts 5669776, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017384, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017384, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00800)
[Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050562, ts 5669936, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017385, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP RFC2833 from   10.200.10.5:20020 (type 101, seq 017385, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00960)
[Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to  10.200.10.5:20020 (type 18, seq 050563, ts 5670096, len 20)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  RTP packet from10.200.10.5:20020 (type 101, seq 017386, ts 5727107, len 04)
[Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got  R

[asterisk-users] libpri 1.4.11.3 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of version 1.4.11.3 of
libpri. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/

This release fixes a regression in the calling number assignment logic:

  * Calling Number assignment logic change in libpri 1.4.11. Restored the old
behaviour if there is more than one calling number in the incoming SETUP
message.  A network provided number is reported as ANI.
(Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)

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

http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Codec negotiation

[mike mosier]

>From what I have seen if your sip provider does not take g722 then you will
have problems with outgoing calls. When I tried this, the same way you did,
I could make calles externally but had no audio each way reguardless of what
I tried to pass to the sip provider. Best bet is to use what your sip
provider can use or find another provider that that can do g722. That's what
I did when I wanted to use g726.

my2cents

On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys  wrote:

> Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
>
> Regards,
> Mindaugas Kezys
>
> Kolmisoft UAB
> VoIP Billing Solutions
> e-mail: i...@kolmisoft.com
> URL: http://www.kolmisoft.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
> Sent: Tuesday, June 29, 2010 7:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Codec negotiation
>
> On 26 June 2010 22:08, Ryan Wagoner  wrote:
> > I have Polycom phones that support the g722 codec. Adding allow=g722
> > to the [general] section of sip.conf works great and I can make calls
> > between the phones using g722. However Asterisk is negotiating g722
> > for calls going out my voip provider and transcoding these to ulaw. In
> > sip.conf for the provider I have deny=all and allow=ulaw. This can
> > cause potential audio degrading and wastes cpu cycles. If Asterisk
> > knows the trunk only supports ulaw why would it offer g722 to the
> > phone.
> >
> > Ryan
>
> Because the codec is already chosen before the call is made, and you
> told it that g722 is permitted.
>
> There are all sorts of discussions in play about codec negotiation,
> but at this point in time, if you want different behaviour you'll need
> to go and code it yourself, and cross-channeltype this is not going to
> be trivial :)
>
> Cheers,
> Steve
>
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> _
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[asterisk-users] Migrating from key system to asterisk

[Brian Kolaci]

I currently have 4 lines coming into the house.  We currently have an Avaya 
standard analog key system which has served us well, but running extensions is 
a major pain and requires a dedicated run per extension.  I have ethernet run 
throughout the house though.

The first two lines are "home" lines, however line1 I use for my consulting 
during the day, and line2 is used by my wife for her work during the day.  
Line3 is a separate business that nobody else in the household should be 
picking up, however I would like to be able to pick it up from any extension if 
required.  Line4 is a fax, but is used for outgoing calls (like conference 
calls, etc.) so that the inbound lines can be left free.

With the current system, I can choose which outbound trunk is used by just 
selecting the line button I prefer.  My wife just picks up the phone for an 
outgoing call and it will choose line2, and if busy then line1.  She would 
never use line3 or line4.  The kids should never touch lines 3 and 4 either.  
They all know enough to only answer lines 1 and 2 and ignore lines 3 and 4, and 
can tell since each line has a different ring as well as the line lights that 
blink.  We also regularly need to be able to "barge" in on each other's calls 
converting a regular call into a conference.

I know what I'm looking for is SLA (Shared Line Appearance), and I've been 
trying to get that working for a couple of weeks.  It supposedly works 
somewhat, but I cannot find enough coherent documentation as to exactly how to 
set up each of the extensions and the physical Polycom phones.  I do see 
there's still bugs being worked on (even today) thats being tested:  
https://issues.asterisk.org/view.php?id=11688

So there's two questions here.  First, is there a way to accomplish the 
requirements I have above to a generic asterisk installation, or are we talking 
a very large and complicated configuration?  Second, is there anywhere I can 
find some detailed documentation on configuring SLA on Polycom phones?  (For 
example, for each line on the Polycom phone, does it log in with an extension 
number or specifically the string "station1_line1", and does the sip.conf get a 
numbered extension for each line on the phone, or do you specifically put 
[statsion1_line1] in there?)

Thanks in advance...


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Re: [asterisk-users] Codec negotiation

[Mindaugas Kezys]

Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Tuesday, June 29, 2010 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec negotiation

On 26 June 2010 22:08, Ryan Wagoner  wrote:
> I have Polycom phones that support the g722 codec. Adding allow=g722
> to the [general] section of sip.conf works great and I can make calls
> between the phones using g722. However Asterisk is negotiating g722
> for calls going out my voip provider and transcoding these to ulaw. In
> sip.conf for the provider I have deny=all and allow=ulaw. This can
> cause potential audio degrading and wastes cpu cycles. If Asterisk
> knows the trunk only supports ulaw why would it offer g722 to the
> phone.
>
> Ryan

Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.

There are all sorts of discussions in play about codec negotiation,
but at this point in time, if you want different behaviour you'll need
to go and code it yourself, and cross-channeltype this is not going to
be trivial :)

Cheers,
Steve

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[asterisk-users] Strange Asterisk/SIP call forwarding behavior

[Myles Wakeham]

I have a small Asterisk 1.4.2 system that I run out of my home based 
business, and my Dialplan has it set to send any incoming call to my 
desk Grandstream phone.  Works really well.

When I leave the office, I need to re-direct the calls to my cell phone. 
  I tried to do this through my Grandstream phone, but it really didn't 
work all that well.  Callers would tell me that they tried to call but 
got cut off, etc.

So I decided to setup my dialplan with a variable in it for the 
destination phone to route to (SIP/MyDeskPhone or Local/15) 
(number excluded for obvious readons) to my cell phone.

What I'm finding, however, is that my Dialplan has about a 50% chance of 
successfully routing the phone call through to my cell phone when I'm 
switched over to using it.  I suspect it has something to do with a 
timing issue with my outgoing SIP provider or something like that.

What I'm looking for is some sort of advice on 'best practice' to handle 
call routing to my cell phone vs. keeping it on my LAN to my desktop 
phone.  As I mentioned, the desktop phone works flawlessly and I'm 
trying to get the same results on my cell phone so I don't lose any 
calls in the process.

Thanks in advance for any advice.

Myles



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Re: [asterisk-users] peer IP address in CDR

[Remco Bressers]

I'll try it out tomorrow. 

Youre my hero of the day!

Regards,

Remco


Op 29 jun. 2010 om 17:45 heeft Zeeshan Zakaria  het 
volgende geschreven:

> Just put exten => _,s, before the MYSQL ...
> 
> Zeeshan A Zakaria
> 
> --
> www.ilovetovoip.com
> 
>> On 2010-06-29 11:41 AM, "Remco Bressers"  wrote:
>> 
>> Thanks Zeeshan, but i don't use (and understand) AEL :)
>> 
>> Any regular examples out there? :)
>> 
>> regards,
>> 
>> Remco
>> 
>> 
>> On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
>> > Let me make it simple for you:
>> > 
>> > Add a column ...
>> 
>> > mailto:list-aster...@skycomuk.com>> wrote:
>> > 
>> > Remco Bressers wr...
>> 
>> --
>> _
>> -- Bandwidth and Colocation Pr...
>> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
> 
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Re: [asterisk-users] Find a way to block brute force attacks.

[Rodrigo Lang]

Thanks again.

But it was a question pending. It's possible AMI show failure resgisters and
wrong password? Because I already have a Java program for AMI and a few
lines of modification would solve my problem if asterisk sends the
information to the AMI.



Thanks,
Rodrigo Lang.



2010/6/29 Andrew Latham 

> Please start here http://www.spamhaus.org/drop/ with your BGP
> routes   Then move up to log parsing.
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
> * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
> * Learn more about Linux http://en.wikipedia.org/wiki/Linux
> * Learn more about Tux http://en.wikipedia.org/wiki/Tux
>
>
>
> On Tue, Jun 29, 2010 at 1:38 PM, Zeeshan Zakaria 
> wrote:
> > If I didn't have fail2ban, I would have way over 20k of these entries in
> my
> > asterisk log.
> >
> > Zeeshan A Zakaria
> >
> > --
> > www.ilovetovoip.com
> >
> > On 2010-06-29 1:36 PM, "Rodrigo Lang" 
> wrote:
> >
> > Good afternoon.
> >
> > Thanks to everyone for answers. What I find strange is the asterisk does
> not
> > have any native tool for him to SIP server security. Here's an example of
> > the syslog messages from asterisk:
> >
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> > [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> > ' failed for '116 .124.128.82 '- Wrong password
> >
> > From what I told there is around twenty thousand records that at one
> time.
> > And at least once a week I receive such an attack coming from a different
> > ip.
> >
> > I will read the articles. Thanks again to everyone.
> >
> >
> > Regards,
> > Rodrigo Lang.
> >
> >
> > 2010/6/29 Kenny Watson 
> >
> >>
> >> Hi, you can use fail2ban
> >> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asteri.
> ..
> >
> > --
> > _
> > -- 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
> >
> > --
> > _
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> >
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> > To UNSUBSCRIBE or update options visit:
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> >
>
> --
> _
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Re: [asterisk-users] Find a way to block brute force attacks.

[Andrew Latham]

Please start here http://www.spamhaus.org/drop/ with your BGP
routes   Then move up to log parsing.


~
Andrew "lathama" Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Tue, Jun 29, 2010 at 1:38 PM, Zeeshan Zakaria  wrote:
> If I didn't have fail2ban, I would have way over 20k of these entries in my
> asterisk log.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-06-29 1:36 PM, "Rodrigo Lang"  wrote:
>
> Good afternoon.
>
> Thanks to everyone for answers. What I find strange is the asterisk does not
> have any native tool for him to SIP server security. Here's an example of
> the syslog messages from asterisk:
>
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
> [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
> ' failed for '116 .124.128.82 '- Wrong password
>
> From what I told there is around twenty thousand records that at one time.
> And at least once a week I receive such an attack coming from a different
> ip.
>
> I will read the articles. Thanks again to everyone.
>
>
> Regards,
> Rodrigo Lang.
>
>
> 2010/6/29 Kenny Watson 
>
>>
>> Hi, you can use fail2ban
>> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asteri...
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
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>
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Re: [asterisk-users] Find a way to block brute force attacks.

[Zeeshan Zakaria]

If I didn't have fail2ban, I would have way over 20k of these entries in my
asterisk log.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-29 1:36 PM, "Rodrigo Lang"  wrote:

Good afternoon.

Thanks to everyone for answers. What I find strange is the asterisk does not
have any native tool for him to SIP server security. Here's an example of
the syslog messages from asterisk:

[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password

>From what I told there is around twenty thousand records that at one time. And
at least once a week I receive such an attack coming from a different ip.

I will read the articles. Thanks again to everyone.


Regards,
Rodrigo Lang.


2010/6/29 Kenny Watson 


>
> Hi, you can use fail2ban
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asteri...


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Re: [asterisk-users] Find a way to block brute force attacks.

[Rodrigo Lang]

Good afternoon.

Thanks to everyone for answers. What I find strange is the asterisk does not
have any native tool for him to SIP server security. Here's an example of
the syslog messages from asterisk:

[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.128.82 '- Wrong password

>From what I told there is around twenty thousand records that at one time. And
at least once a week I receive such an attack coming from a different ip.

I will read the articles. Thanks again to everyone.


Regards,
Rodrigo Lang.


2010/6/29 Kenny Watson 

> Hi, you can use fail2ban
> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
>
> Which works well, when a pattern is found in a log file it addes in an
> iptables rules to block the traffic for a period.
>
> On debian you can apt-get install fail2ban and on centos/redhat yum -i
> fail2ban
>
> Thanks
>
> Kenny
>
> - Original Message -
> From: "Gareth Blades" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> Sent: Tuesday, 29 June, 2010 4:12:42 PM
> Subject: Re: [asterisk-users] Find a way to block brute force attacks.
>
> Rodrigo Lang wrote:
> > Hello list.
> >
> > I'm trying to find a way to block any ip that tries to login more than
> > three times with the wrong password and try to log in three different
> > extensions. For I have suffered some brute force attacks on my asterisk
> > in the morning period.
> >
> > The idea would be: Any ip with three attempts without success to log
> > into an extension is blocked.
> >
> > Is there any way to accomplish this directly by the asterisk? Or is
> > there some kind of asterisk spit this information via the AMI?
> >
> > I was wondering to make a Java program to listen to the AMI and create a
> > rule in iptables for ip in specific.
> >
> > Does anyone have any suggestions?
> >
> >
> > Thanks,
> > Rodrigo Lang.
> >
> Does asterisk log the failed attempts to a file?
> If so then you could use sshblack to monitor the file for incorrect
> logins. It will add firewalls rules to a custom iptables chain based on
> various criteria. You can then point incoming SIP connections through
> this chain so offenders will be forewalled for a specific amount of time.
> http://www.pettingers.org/code/sshblack.html
>
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Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9

[Carlos Chavez]

On Tue, 2010-06-29 at 10:04 +0800, Zhang Shukun wrote:
> hi, list
>  i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
> 
> i want to use CentOS5.2 or CentOS 5.4.  Which is better and stable?
> Thanks for your help.
> 
> 
The best OS is the one you are most confortable and the one you have
the most experience with.

CentOS is now on version 5.5, you can start with any CentOS 5 and when
you update it it will be 5.5.  Always make sure your OS has all the
security updates applied.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Codec negotiation

[Steve Davies]

On 26 June 2010 22:08, Ryan Wagoner  wrote:
> I have Polycom phones that support the g722 codec. Adding allow=g722
> to the [general] section of sip.conf works great and I can make calls
> between the phones using g722. However Asterisk is negotiating g722
> for calls going out my voip provider and transcoding these to ulaw. In
> sip.conf for the provider I have deny=all and allow=ulaw. This can
> cause potential audio degrading and wastes cpu cycles. If Asterisk
> knows the trunk only supports ulaw why would it offer g722 to the
> phone.
>
> Ryan

Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.

There are all sorts of discussions in play about codec negotiation,
but at this point in time, if you want different behaviour you'll need
to go and code it yourself, and cross-channeltype this is not going to
be trivial :)

Cheers,
Steve

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[asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs

[Kenny Watson]

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. 

I have g729 and alaw trunks from a pbx /sip providers. 


The howler screamer will only transcode from g729 to alaw / ulaw but my voice 
prompts are in SLIN and throws errors when i try and access these applications. 


Is it simply a case of converting the prompts into other codecs and asterisk 
will pick these up? 

  

Thanks 


Kenny 
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Re: [asterisk-users] Call file structure and syntax

[Mike Ely]

Yep, I saw that and it's just not the case.  I was having it dial my desk
extension, which was decidedly not busy at the time...


On 6/28/10 5:30 PM, "Philipp von Klitzing"
 wrote:

>> Well, I¹ve tried this, and something just isn¹t right.
> 
> Look here:
> 
>> Event: Hangup
>> Channel: SIP/ShoreTel-1-0004
>> Cause: 17   
>> Cause-txt: User busy
> 


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Re: [asterisk-users] peer IP address in CDR

[Zeeshan Zakaria]

Just put exten => _,s, before the MYSQL ...

Zeeshan A Zakaria

--
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On 2010-06-29 11:41 AM, "Remco Bressers"  wrote:

Thanks Zeeshan, but i don't use (and understand) AEL :)

Any regular examples out there? :)

regards,

Remco



On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
> Let me make it simple for you:
>
> Add a column ...

> mailto:list-aster...@skycomuk.com>> wrote:
>
> Remco Bressers wr...
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Re: [asterisk-users] peer IP address in CDR

[Remco Bressers]

Thanks Zeeshan, but i don't use (and understand) AEL :)

Any regular examples out there? :)

regards,

Remco


On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
> Let me make it simple for you:
> 
> Add a column to your table, e.g. `my column`.
> 
> In the dialplan do the following (AEL example):
> 
> MYSQL(Connect connid localhost username password database);
> MYSQL(Query resultid ${connid} INSERT INTO `cdr`
> (`mycolumn`) VALUES('${SIPCHANINFO(ip)}'));
> MYSQL(Disconnect ${connid});
> 
> --
> Zeeshan
> 
> On Tue, Jun 29, 2010 at 10:32 AM, Gareth Blades
> mailto:list-aster...@skycomuk.com>> wrote:
> 
> Remco Bressers wrote:
> > Hi,
> >
> > Sorry, but i forgot to notice that i am already using the 'userfield'
> > column so that's not a possibility. Is there any way i can add the IP
> > address to a custom MySQL field in CDR? With AGI possibly? The problem
> > is, that the CDR entry is written in MySQL when the call is
> hungup, so i
> > have no possibility to write the IP address after a call.
> >
> > Regards,
> >
> > Remco
> >
> 
> See my earlier post. You can certenly write the information after the
> call is hung up by using the 'h' extension. I do this myself to write
> the calculated call cost to a custom column in the mysql table.
> 
> --
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> 
> 
> 
> 
> -- 
> Zeeshan A Zakaria
> 

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Re: [asterisk-users] peer IP address in CDR

[Zeeshan Zakaria]

Let me make it simple for you:

Add a column to your table, e.g. `my column`.

In the dialplan do the following (AEL example):

MYSQL(Connect connid localhost username password database);
MYSQL(Query resultid ${connid} INSERT INTO `cdr` (`mycolumn`)
VALUES('${SIPCHANINFO(ip)}'));
MYSQL(Disconnect ${connid});

--
Zeeshan

On Tue, Jun 29, 2010 at 10:32 AM, Gareth Blades
wrote:

> Remco Bressers wrote:
> > Hi,
> >
> > Sorry, but i forgot to notice that i am already using the 'userfield'
> > column so that's not a possibility. Is there any way i can add the IP
> > address to a custom MySQL field in CDR? With AGI possibly? The problem
> > is, that the CDR entry is written in MySQL when the call is hungup, so i
> > have no possibility to write the IP address after a call.
> >
> > Regards,
> >
> > Remco
> >
>
> See my earlier post. You can certenly write the information after the
> call is hung up by using the 'h' extension. I do this myself to write
> the calculated call cost to a custom column in the mysql table.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Re: [asterisk-users] Find a way to block brute force attacks.

[Kenny Watson]

Hi, you can use fail2ban 
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk

Which works well, when a pattern is found in a log file it addes in an iptables 
rules to block the traffic for a period.

On debian you can apt-get install fail2ban and on centos/redhat yum -i fail2ban

Thanks

Kenny

- Original Message -
From: "Gareth Blades" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, 29 June, 2010 4:12:42 PM
Subject: Re: [asterisk-users] Find a way to block brute force attacks.

Rodrigo Lang wrote:
> Hello list.
> 
> I'm trying to find a way to block any ip that tries to login more than 
> three times with the wrong password and try to log in three different 
> extensions. For I have suffered some brute force attacks on my asterisk 
> in the morning period.
> 
> The idea would be: Any ip with three attempts without success to log 
> into an extension is blocked.
> 
> Is there any way to accomplish this directly by the asterisk? Or is 
> there some kind of asterisk spit this information via the AMI?
> 
> I was wondering to make a Java program to listen to the AMI and create a 
> rule in iptables for ip in specific.
> 
> Does anyone have any suggestions?
> 
> 
> Thanks,
> Rodrigo Lang.
> 
Does asterisk log the failed attempts to a file?
If so then you could use sshblack to monitor the file for incorrect 
logins. It will add firewalls rules to a custom iptables chain based on 
various criteria. You can then point incoming SIP connections through 
this chain so offenders will be forewalled for a specific amount of time.
http://www.pettingers.org/code/sshblack.html

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Re: [asterisk-users] peer IP address in CDR

[Philipp von Klitzing]

Hi!

> Do you already have script to capture user's IP address? If not, check
> it here how I am capturing it:
>
> http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
> within-the-dialplan

Or simply use one fo these:

  ${SIPCHANINFO(peerip)}
  ${SIPCHANINFO(recvip)}
  ${SIPCHANINFO(uri)}

More details with "show function SIPCHANINFO" on the CLI.

But: Anyone has an idea how to store the codec(s) that were employed for
the call in the CDR (or access it during hangup in the dialplan)?

The Wiki has a suggested patch to enhance SIPCHANINFO, but I wonder if 
there is a cleaner and built-in way to do it:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo

Philipp


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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Zeeshan Zakaria]

Actually putty does it all. I don't know which putty you are using, maybe
try downloading it again and explore its settings.

Zeeshan A Zakaria

--
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On 2010-06-29 11:12 AM, "Roderick A. Anderson" 
wrote:

On 06/29/2010 06:53 AM, bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very...
I haven't used Putty for several months and that was with a setup I'd
made several years ago so I can't, off the top of my head, tell you how
I did it; but I had the remote system's name or IP in the window title
bar.  It might nave been the name I saved the connection as.

Look in the configuration under Terminal.  Something like a %s and make
sure the terminal type is either Linux or ANSI.  Again too long ago.


Rod
--

>
> Can you please weigh in and tell me what your favorite terminal software
> is and why?
>
> Thank...

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Re: [asterisk-users] Find a way to block brute force attacks.

[Gareth Blades]

Rodrigo Lang wrote:
> Hello list.
> 
> I'm trying to find a way to block any ip that tries to login more than 
> three times with the wrong password and try to log in three different 
> extensions. For I have suffered some brute force attacks on my asterisk 
> in the morning period.
> 
> The idea would be: Any ip with three attempts without success to log 
> into an extension is blocked.
> 
> Is there any way to accomplish this directly by the asterisk? Or is 
> there some kind of asterisk spit this information via the AMI?
> 
> I was wondering to make a Java program to listen to the AMI and create a 
> rule in iptables for ip in specific.
> 
> Does anyone have any suggestions?
> 
> 
> Thanks,
> Rodrigo Lang.
> 
Does asterisk log the failed attempts to a file?
If so then you could use sshblack to monitor the file for incorrect 
logins. It will add firewalls rules to a custom iptables chain based on 
various criteria. You can then point incoming SIP connections through 
this chain so offenders will be forewalled for a specific amount of time.
http://www.pettingers.org/code/sshblack.html

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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Danny Nicholas]

I use PUTTY 0.58 and have Window title and scroll control for 20K+ lines.
It could use some improvements, but it is more than adequate for "green
screen" control.  The quality of Putty and many other applications depends
on how you choose to control it.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
Anderson
Sent: Tuesday, June 29, 2010 10:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] What TERMINAL software do you use for MS
Windows platform and WHY?

On 06/29/2010 06:53 AM, bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very nice as in it allows many many
> SSH profiles to be saved and allows tunneling etcbut it's not very
> good when it comes to scrolling up and down, colors, text size, and
> specially it doesn't give a title to the opened instance. Maybe giving
> the IP address as the title of the window would help a lot if you have
> many different servers opened at the same time.

I haven't used Putty for several months and that was with a setup I'd 
made several years ago so I can't, off the top of my head, tell you how 
I did it; but I had the remote system's name or IP in the window title 
bar.  It might nave been the name I saved the connection as.

Look in the configuration under Terminal.  Something like a %s and make 
sure the terminal type is either Linux or ANSI.  Again too long ago.


Rod
-- 
>
> Can you please weigh in and tell me what your favorite terminal software
> is and why?
>
> Thanks,
> Bruce
>


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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Roderick A. Anderson]

On 06/29/2010 06:53 AM, bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very nice as in it allows many many
> SSH profiles to be saved and allows tunneling etcbut it's not very
> good when it comes to scrolling up and down, colors, text size, and
> specially it doesn't give a title to the opened instance. Maybe giving
> the IP address as the title of the window would help a lot if you have
> many different servers opened at the same time.

I haven't used Putty for several months and that was with a setup I'd 
made several years ago so I can't, off the top of my head, tell you how 
I did it; but I had the remote system's name or IP in the window title 
bar.  It might nave been the name I saved the connection as.

Look in the configuration under Terminal.  Something like a %s and make 
sure the terminal type is either Linux or ANSI.  Again too long ago.


Rod
-- 
>
> Can you please weigh in and tell me what your favorite terminal software
> is and why?
>
> Thanks,
> Bruce
>


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Re: [asterisk-users] Find a way to block brute force attacks.

[Mark R]

There are some good suggestions here as a starting point:

http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/

Rgds,

mcr

On 29 June 2010 15:39, Rodrigo Lang  wrote:

> Hello list.
>
> I'm trying to find a way to block any ip that tries to login more than
> three times with the wrong password and try to log in three different
> extensions. For I have suffered some brute force attacks on my asterisk in
> the morning period.
>
> The idea would be: Any ip with three attempts without success to log into
> an extension is blocked.
>
> Is there any way to accomplish this directly by the asterisk? Or is there
> some kind of asterisk spit this information via the AMI?
>
> I was wondering to make a Java program to listen to the AMI and create a
> rule in iptables for ip in specific.
>
> Does anyone have any suggestions?
>
>
> Thanks,
> Rodrigo Lang.
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[William Stillwell (Lists)]

I use SecureCRT+FX , and use ansi graphics.

Putty is nice w/WinSCP as well.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 29, 2010 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What TERMINAL software do you use for MS
Windows platform and WHY?

bruce bruce wrote:
> Hi Everyone,
> 
> I am accustomed to PUTTY and it's very nice as in it allows many many 
> SSH profiles to be saved and allows tunneling etcbut it's not very 
> good when it comes to scrolling up and down, colors, text size, and 
> specially it doesn't give a title to the opened instance. Maybe giving 
> the IP address as the title of the window would help a lot if you have 
> many different servers opened at the same time.
> 
> Can you please weigh in and tell me what your favorite terminal software 
> is and why?
> 
> Thanks,
> Bruce
> 

I just run Linux as my desktop and that makes it much easier :)

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[asterisk-users] Find a way to block brute force attacks.

[Rodrigo Lang]

Hello list.

I'm trying to find a way to block any ip that tries to login more than three
times with the wrong password and try to log in three different extensions. For
I have suffered some brute force attacks on my asterisk in the morning
period.

The idea would be: Any ip with three attempts without success to log into an
extension is blocked.

Is there any way to accomplish this directly by the asterisk? Or is there
some kind of asterisk spit this information via the AMI?

I was wondering to make a Java program to listen to the AMI and create a
rule in iptables for ip in specific.

Does anyone have any suggestions?


Thanks,
Rodrigo Lang.
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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Niccolò Belli]

I do not use windows on the desktop/laptop, but when I have to I use putty.

Darkbasic

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Re: [asterisk-users] peer IP address in CDR

[Gareth Blades]

Remco Bressers wrote:
> Hi,
> 
> Sorry, but i forgot to notice that i am already using the 'userfield'
> column so that's not a possibility. Is there any way i can add the IP
> address to a custom MySQL field in CDR? With AGI possibly? The problem
> is, that the CDR entry is written in MySQL when the call is hungup, so i
> have no possibility to write the IP address after a call.
> 
> Regards,
> 
> Remco
> 

See my earlier post. You can certenly write the information after the 
call is hung up by using the 'h' extension. I do this myself to write 
the calculated call cost to a custom column in the mysql table.

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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Luis Morales]

I like putty too. There are many features included in this client, for
example an freindly interface to setup tunnels, X11, and another
features.

Take a look into putty website.

http://the.earth.li/~sgtatham/putty/0.60/htmldoc/


Regards,


On Tue, Jun 29, 2010 at 9:52 AM, Tzafrir Cohen  wrote:
> On Tue, Jun 29, 2010 at 09:53:42AM -0400, bruce bruce wrote:
>> Hi Everyone,
>>
>> I am accustomed to PUTTY and it's very nice as in it allows many many SSH
>> profiles to be saved and allows tunneling etcbut it's not very good when
>> it comes to scrolling up and down, colors, text size, and specially it
>> doesn't give a title to the opened instance. Maybe giving the IP address as
>> the title of the window would help a lot if you have many different servers
>> opened at the same time.
>>
>> Can you please weigh in and tell me what your favorite terminal software is
>> and why?
>
> Not that I use it myself, but I get recommendations for
> http://code.google.com/p/msysgit/
> as a rather usable distribution of terminal+ssh .
>
> --
>               Tzafrir Cohen
> icq#16849755              jabber:tzafrir.co...@xorcom.com
> +972-50-7952406           mailto:tzafrir.co...@xorcom.com
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Re: [asterisk-users] peer IP address in CDR

[Remco Bressers]

Hi,

Sorry, but i forgot to notice that i am already using the 'userfield'
column so that's not a possibility. Is there any way i can add the IP
address to a custom MySQL field in CDR? With AGI possibly? The problem
is, that the CDR entry is written in MySQL when the call is hungup, so i
have no possibility to write the IP address after a call.

Regards,

Remco

On 06/29/2010 03:53 PM, Zeeshan Zakaria wrote:
> Hi,
> 
> There is usually an empty column in the cdr table named 'userfield'. You
> can also add a column of your own. Then in the dialplan use:
> 
> Set(CDR(userfield)="user IP address")
> 
> And this will automatically add this information into the userfield column.
> 
> Do you already have script to capture user's IP address? If not, check
> it here how I am capturing it:
> 
> http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-within-the-dialplan
> 
> Zeeshan A Zakaria
> 
> --
> www.ilovetovoip.com 
> 
>> On 2010-06-29 8:20 AM, "Faisal Hanif" > > wrote:
>>
>> Simply set it to costume field of cdrs in dialplan and you will have
>> it a part of native cdr
>> Regards,
>>
>> *Faisal Hanif*
>>
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 


-- 
Met vriendelijke groet,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl
altijd online? www.signet.nl

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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Tzafrir Cohen]

On Tue, Jun 29, 2010 at 09:53:42AM -0400, bruce bruce wrote:
> Hi Everyone,
> 
> I am accustomed to PUTTY and it's very nice as in it allows many many SSH
> profiles to be saved and allows tunneling etcbut it's not very good when
> it comes to scrolling up and down, colors, text size, and specially it
> doesn't give a title to the opened instance. Maybe giving the IP address as
> the title of the window would help a lot if you have many different servers
> opened at the same time.
> 
> Can you please weigh in and tell me what your favorite terminal software is
> and why?

Not that I use it myself, but I get recommendations for
http://code.google.com/p/msysgit/
as a rather usable distribution of terminal+ssh .

-- 
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[Gareth Blades]

bruce bruce wrote:
> Hi Everyone,
> 
> I am accustomed to PUTTY and it's very nice as in it allows many many 
> SSH profiles to be saved and allows tunneling etcbut it's not very 
> good when it comes to scrolling up and down, colors, text size, and 
> specially it doesn't give a title to the opened instance. Maybe giving 
> the IP address as the title of the window would help a lot if you have 
> many different servers opened at the same time.
> 
> Can you please weigh in and tell me what your favorite terminal software 
> is and why?
> 
> Thanks,
> Bruce
> 

I just run Linux as my desktop and that makes it much easier :)

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Re: [asterisk-users] peer IP address in CDR

[Gareth Blades]

Remco Bressers wrote:
> Hi,
> 
> The subject says it all. Is it possible to put the IP address of the
> peer in the CDR records? Using AGI maybe?
> 

Yes you can either put the information in the userfield if you are using 
a plain text file.

If you are storing to a mysql table for example then you can create 
additional columns and write whatever information you want to them. See 
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql for an example of 
how to do this.

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Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9

[Tiago Geada]

Ubuntu is not Debian.

I would recommend Debian tho, its rock solid and it jsut works (for
anything)

On 29 June 2010 12:29, Paul Belanger  wrote:

> On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun  wrote:
> > i want to know what is the best OS for install Asterisk 1.6.2.9,
> > which should work properly on working system.
> >
> Ubuntu 10.04 Server ?
>
> --
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> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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[asterisk-users] SIP Delay with remote stations?

[William Stillwell (Lists)]

I have several remote phones that experience a slight "call" delay when
answering phones, ie, they will answer, speak a few words, and then the
remote caller will hear them, and the first half is cutoff?

 

Any idea what could be causing this?

 

 

Thanks,

Bill.

 

 

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Re: [asterisk-users] cmd Authenticate

[Danny Nicholas]

AFAIK, this will only address successful authentications.  I think the OP
wanted to be able to know what the user had entered on failed attempts.
Since I've added another layer to this onion, I think the best option is to
use Read followed by an AGI if you have bells and whistles like this and a
Gotoif if you are doing simple stuff.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Tuesday, June 29, 2010 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cmd Authenticate

Hi!

> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?

Use option a of Authenticate together with ${CDR(accountcode)}

Philipp


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Re: [asterisk-users] cmd Authenticate

[Zeeshan Zakaria]

Due to this reason I am doing authentications using Read().

Zeeshan A Zakaria

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On 2010-06-29 9:36 AM, "Coco Richard"  wrote:

Danny, Doug

thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().

rich


On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle  wrote:
> Coco Richard wrote:
>> Hi...
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Re: [asterisk-users] cmd Authenticate

[Philipp von Klitzing]

Hi!

> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?

Use option a of Authenticate together with ${CDR(accountcode)}

Philipp


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Re: [asterisk-users] peer IP address in CDR

[Zeeshan Zakaria]

Hi,

There is usually an empty column in the cdr table named 'userfield'. You can
also add a column of your own. Then in the dialplan use:

Set(CDR(userfield)="user IP address")

And this will automatically add this information into the userfield column.

Do you already have script to capture user's IP address? If not, check it
here how I am capturing it:

http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-within-the-dialplan

Zeeshan A Zakaria

--
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On 2010-06-29 8:20 AM, "Faisal Hanif"  wrote:

 Simply set it to costume field of cdrs in dialplan and you will have it a
part of native cdr
 Regards,

*Faisal Hanif*

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[asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[bruce bruce]

Hi Everyone,

I am accustomed to PUTTY and it's very nice as in it allows many many SSH
profiles to be saved and allows tunneling etcbut it's not very good when
it comes to scrolling up and down, colors, text size, and specially it
doesn't give a title to the opened instance. Maybe giving the IP address as
the title of the window would help a lot if you have many different servers
opened at the same time.

Can you please weigh in and tell me what your favorite terminal software is
and why?

Thanks,
Bruce
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Re: [asterisk-users] cmd Authenticate

[Coco Richard]

Danny, Doug

thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().

rich

On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle  wrote:
> Coco Richard wrote:
>> Hi,
>>
>> i need to save into a local variable the user's input dialed during
>> the cmd Authenticate(). Is there a way to do it?
>>
>>
>
> core show application authenticate
> hylafax*CLI>
>   -= Info about application 'Authenticate' =-
>
> [Synopsis]
> Authenticate a user
>
>
>   Options:
>      a - Set the channels' account code to the password that is entered
>
> --
>
> You probably could use option a.
>
> But, I'd suggest that instead of using authenticate, you code something
> using the read option.
>
> I use read to authenticate conference administration.
>
> [check-password]
>
> exten => s,1,Read(get-admin-password|enter-password|||3|)
> exten => s,n,Gotoif($["${LEN(${get-admin-password})}" < "1"]?9:3)
> exten => s,n, some dialplan magic here.
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
>
>
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Re: [asterisk-users] cmd Authenticate

[Doug Lytle]

Coco Richard wrote:
> Hi,
>
> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?
>
>

core show application authenticate
hylafax*CLI>
   -= Info about application 'Authenticate' =-

[Synopsis]
Authenticate a user


   Options:
  a - Set the channels' account code to the password that is entered

--

You probably could use option a.

But, I'd suggest that instead of using authenticate, you code something 
using the read option.

I use read to authenticate conference administration.

[check-password]

exten => s,1,Read(get-admin-password|enter-password|||3|)
exten => s,n,Gotoif($["${LEN(${get-admin-password})}" < "1"]?9:3)
exten => s,n, some dialplan magic here.

Doug


-- 

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"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Is Centos 64 bit or 32 bit better?

[Luis Morales]

Depend of your hardware. For example if you plan use 8G or more in RAM
it's better choice 64Bits distro.  There are others benefits for
example the size on databases, logs files, memory use, recording
files, etc.


Regards,


On Tue, Jun 29, 2010 at 8:38 AM, Steve Underwood  wrote:
> On 06/29/2010 05:35 PM, Gareth Blades wrote:
>> Zhang Shukun wrote:
>>
>>> hi, all
>>>          after a long time development, i need to deploy a production 
>>> system.
>>>
>>> i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing 
>>> confused me.
>>>
>>> my computer hardware support 64 bit OS.
>>>
>>> my question is : should i use Centos 5.4 64bit or  Centos 5.4 32bit?
>>>
>>> which is better for my asterisk ? consider compatibilityand stability.
>>>
>>> this is a new machine , only used for asterisk, no other apps.
>>>
>>> Thank you in advance!
>>>
>> 64bit will give you more adressible memory and faster performance when
>> dealing with 64bit numbers. Neither of these will really give you any
>> benefit but Asterisk and all the extras I have seen all work fine on
>> 64bit so there is no real reason not to go for it.
>>
>>
> Actually most DSP code runs *much* faster on a 64 bit machine. I think
> it mostly the better register set which results in that.
>
> Steve
>
>
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Re: [asterisk-users] cmd Authenticate

[Danny Nicholas]

I believe that the information keyed is just trashed after authentication.
You could modify app_authenticate.c to set a variable with the passed
information (this might already be included in the 1.6/1.8 branches, I just
deal with 1.4).  Your other option would be to use a read/gotoif pair in
place of authenticate.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Coco Richard
Sent: Tuesday, June 29, 2010 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] cmd Authenticate

Hi,

i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?

thx
rich

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[asterisk-users] OrderlyStats SE 1.6.2l now available.

[Matt King]

Hello,

Just thought you might like to know that our popular call center 
management and statistics package, OrderlyStats SE, has just got a new 
release.

Version 1.6.2l includes a several configuration changes and enhancements 
to provide seamless call integration with the popular Elastix 
distribution of Asterisk.

To download your free evaluation version, and start benefiting from 
improved visibility and easy call center management today, just visit 
http://www.orderlyq.com/asteriskcallcenterstatistics.html

Thanks for reading.

Kind regards,

Matt King
Managing Director
Orderly Software Ltd.

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Re: [asterisk-users] Is Centos 64 bit or 32 bit better?

[Steve Underwood]

On 06/29/2010 05:35 PM, Gareth Blades wrote:
> Zhang Shukun wrote:
>
>> hi, all
>>  after a long time development, i need to deploy a production system.
>>
>> i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused 
>> me.
>>
>> my computer hardware support 64 bit OS.
>>
>> my question is : should i use Centos 5.4 64bit or  Centos 5.4 32bit?
>>
>> which is better for my asterisk ? consider compatibilityand stability.
>>
>> this is a new machine , only used for asterisk, no other apps.
>>
>> Thank you in advance!
>>  
> 64bit will give you more adressible memory and faster performance when
> dealing with 64bit numbers. Neither of these will really give you any
> benefit but Asterisk and all the extras I have seen all work fine on
> 64bit so there is no real reason not to go for it.
>
>
Actually most DSP code runs *much* faster on a 64 bit machine. I think 
it mostly the better register set which results in that.

Steve


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[asterisk-users] cmd Authenticate

[Coco Richard]

Hi,

i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?

thx
rich

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Re: [asterisk-users] Problem with GoToIfTime

[Peder]

 =  
| '-' 
| "*" 

 

It's either a range of days, e.g. 29-30, or * for don't care.  So do 29-30.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, June 29, 2010 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with GoToIfTime

 

Hello list,

why is it that GoToIfTime thinks a date of *|*|29-*|jun is not valid ??


[Jun 29 14:06:34] -- Executing [...@macro-vac:10]
GotoIfTime("SIP/testcorp-0036", "*|*|29-*|jun?onvac") in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*',
assuming none
[Jun 29 14:06:34] -- Executing [...@macro-vac:11]
GotoIfTime("SIP/testcorp-0036", "*|*|*-1|jul?onvac") in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4120 get_range: Invalid day '*',
assuming none


I want to set a period of 29 June till first July.



Jonas.

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[asterisk-users] Problem with GoToIfTime

[Jonas Kellens]

Hello list,

why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ??


[Jun 29 14:06:34] -- Executing [...@macro-vac:10] 
*GotoIfTime*("SIP/testcorp-0036", "**|*|29-*|jun*?onvac") in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day 
'*', assuming none
[Jun 29 14:06:34] -- Executing [...@macro-vac:11] 
*GotoIfTime*("SIP/testcorp-0036", "**|*|*-1|jul*?onvac") in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4120 get_range: Invalid day '*', 
assuming none



I want to set a period of 29 June till first July.



Jonas.
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Re: [asterisk-users] peer IP address in CDR

[Faisal Hanif]

 Simply set it to costume field of cdrs in dialplan and you will have 
it a part of native cdr

Regards,

*Faisal Hanif*
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Re: [asterisk-users] G729 license key registration

[Gordon Henderson]

On Tue, 29 Jun 2010, Kiss András wrote:


2010/6/25 Remco Bressers :

On 06/25/2010 09:48 AM, Kiss András wrote:

You selected 5, G.729 Codec
Please enter your Key-ID: G729-10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: 404 - Key not found.

Any suggestions?


How about contacting Digium about this?


I`ve tried, got no response (yet)
I`ve found a thread on this list with the same problem, without the
solution: 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg219314.html


Dump the digium one and download the "free" one and make sure you don't 
utilise more channels of g729 than you've paid licenses for... (or just 
buy more licenses when you need more channels)


Probably not the technically correct way to do things, but no-one replied 
the last time I suggested this on the list...


Or try a commercially licensed alternative - e.g. 
http://www.howlertech.com/products/howlets/ perhaps their licensing works 
better?


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[asterisk-users] peer IP address in CDR

[Remco Bressers]

Hi,

The subject says it all. Is it possible to put the IP address of the
peer in the CDR records? Using AGI maybe?

-- 
Kind regards,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl

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Re: [asterisk-users] restricting sip users to a certain useragent

[Faisal Hanif]

 Hi,

If you use curl realtime for registrations you can add useragnet check 
in your CGI and also lot of else as well.

Regards,

*Faisal Hanif
*On 6/29/2010 4:48 PM, Tarek Sawah wrote:
well there are two restrictions.. the IP address of the station they 
are using it .. and the UserAgent..
one thing my agents hardly understand Computers .. and their computer 
skills are limited to Microsoft Office products and telemarketing.
i'm not afraid of hackers or cracker .. security is not guaranteed .. 
but i need to restrict the agents to their seats and my CRM software


-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: 
+1 347 562 2308




> From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Tue, 29 Jun 2010 08:45:01 +0100
> Subject: Re: [asterisk-users] restricting sip users to a certain 
useragent

>
> On Tuesday 29 Jun 2010, Tarek Sawah wrote:
> > . is it possible to
> > force the agents (users) to use a certain UserAgent which is the one
> > built-in our system? this way will prevent the agents we are 
restricting

> > them to only be able to dial through the software which is already
> > restricted to their seats in the call center.. but someone might sniff
> > around .. and get the sip username and password assigned to him 
and use it

> > through Zoiper or any other softphone to make calls .
>
> If someone is *that* determined, what will stop them from modifying the
> user-agent string in some Open Source softphone?
>
> --
> AJS
>
> --
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>
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The New Busy is not the too busy. Combine all your e-mail accounts 
with Hotmail. Get busy. 

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Re: [asterisk-users] restricting sip users to a certain useragent

[Tarek Sawah]

well there are two restrictions.. the IP address of the station they are using 
it .. and the UserAgent..one thing my agents hardly understand Computers .. and 
their computer skills are limited to Microsoft Office products and 
telemarketing. i'm not afraid of hackers or cracker .. security is not 
guaranteed .. but i need to restrict the agents to their seats and my CRM 
software

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






> From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Tue, 29 Jun 2010 08:45:01 +0100
> Subject: Re: [asterisk-users] restricting sip users to a certain useragent
> 
> On Tuesday 29 Jun 2010, Tarek Sawah wrote:
> > . is it possible to
> > force the agents (users) to use a certain UserAgent which is the one
> > built-in our system?  this way will prevent the agents we are restricting
> > them to only be able to dial through the software which is already
> > restricted to their seats in the call center.. but someone might sniff
> > around .. and get the sip username and password assigned to him and use it
> > through Zoiper or any other softphone to make calls .
> 
> If someone is *that* determined, what will stop them from modifying the 
> user-agent string in some Open Source softphone?
> 
> -- 
> AJS
> 
> -- 
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Re: [asterisk-users] Hot to configure trunk in asterisk with a2billing.

[Tarek Sawah]

Lets say you did everything as it was mentioned in the tutorial .. then go into 
Asterisk console and issue the command:sip show peer A2BILLINGCREATEDUSER
if you can't find it.. then simply include additional_a2billing_sip.conf  in 
your sip.conf file.Regards
-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






Date: Tue, 29 Jun 2010 13:41:22 +0530
From: alagudr...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hot to configure trunk in asterisk with a2billing.

Hi All,

I am newbie in this asterisk  and a2billing technology . i had successfully 
installed asterisk in my server fedora -8 [server behind NAT/STUN]
i after installation i can able to create users and tested the call  features 
with X-Lite . the was working fine . 


after i installed the A2Billing in my same server with  follow the steps from 
a2billing installation guide.

but u cant access the users from a2billing in asterisk . if i am trying to 
access the username which is created in a2billing it displayed 


"request timeout" somewhere i missed the configuration, please help me to 
resolve this error .

Thanks,

Gokul.
  
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Re: [asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

[Paul Belanger]

On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun  wrote:
>     i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
Ubuntu 10.04 Server ?

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

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Re: [asterisk-users] How to Add IP address to SIP Domain

[Paul Belanger]

On Tue, Jun 29, 2010 at 3:23 AM, DHAVAL INDRODIYA
 wrote:
> is there anyway to resolve it out, Means if SIP wants to send each call to
> 192.168.1.30 , but without entry in /etc/hosts.
>
Setup a DNS server.

-- 
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blog.polybeacon.com

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Re: [asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

[John Novack]

Zhang Shukun wrote:
> hi, list
>   i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
> i want to use CentOS5.2 or CentOS 5.4.  Which is better and stable?
> Thanks for your help.
>
>
>
Somewhat of a religious argument.
CentOS 5.5 is current, and does work well.
You will also find the Debian/Ubuntu camp on the list

Use what you know.
If you don't know either, use CentOS. 32 it is fine for asterisk
Compile from source

Using the above, your Google searches for help will be the most productive

MO

john Novack

-- 

Dog is my Co-pilot


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Re: [asterisk-users] Is Centos 64 bit or 32 bit better?

[Gareth Blades]

Zhang Shukun wrote:
> hi, all
> after a long time development, i need to deploy a production system.
> 
> i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused 
> me.
> 
> my computer hardware support 64 bit OS.
> 
> my question is : should i use Centos 5.4 64bit or  Centos 5.4 32bit?
> 
> which is better for my asterisk ? consider compatibilityand stability.
> 
> this is a new machine , only used for asterisk, no other apps.
> 
> Thank you in advance!

64bit will give you more adressible memory and faster performance when 
dealing with 64bit numbers. Neither of these will really give you any 
benefit but Asterisk and all the extras I have seen all work fine on 
64bit so there is no real reason not to go for it.


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Re: [asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9

[Gareth Blades]

Zhang Shukun wrote:
> hi, list
>  i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
> 
> i want to use CentOS5.2 or CentOS 5.4.  Which is better and stable?
> Thanks for your help.
> 
> 

Whatever system you go for it should have a long maintenance cycle so 
you dont have to rebuild the box because the os is no longer supported.
Centos is fine and you might also want to consider the commercial Redhat 
or Suse equivilents.
If you go for centos get the latest.

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[asterisk-users] Is Centos 64 bit or 32 bit better?

[Zhang Shukun]

hi, all
after a long time development, i need to deploy a production system.

i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused me.

my computer hardware support 64 bit OS.

my question is : should i use Centos 5.4 64bit or  Centos 5.4 32bit?

which is better for my asterisk ? consider compatibilityand stability.

this is a new machine , only used for asterisk, no other apps.

Thank you in advance!
-- 
Thanks for your supporting,
have a nice day.
Sucan

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Re: [asterisk-users] G729 license key registration

[Kiss András]

2010/6/25 Remco Bressers :
> On 06/25/2010 09:48 AM, Kiss András wrote:
>> You selected 5, G.729 Codec
>> Please enter your Key-ID: G729-10D2X----X
>> This product key cannot be registered!  Please verify you entered the
>> correct product key.
>> Server response: 404 - Key not found.
>>
>> Any suggestions?
>
> How about contacting Digium about this?

I`ve tried, got no response (yet)
I`ve found a thread on this list with the same problem, without the
solution: 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg219314.html

- András

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[asterisk-users] Hot to configure trunk in asterisk with a2billing.

[gokulakrishnan]

Hi All,

I am newbie in this asterisk  and a2billing technology . i had successfully
installed asterisk in my server fedora -8 [server behind NAT/STUN]
i after installation i can able to create users and tested the call
features with X-Lite . the was working fine .

after i installed the A2Billing in my same server with  follow the steps
from a2billing installation guide.

but u cant access the users from a2billing in asterisk . if i am trying to
access the username which is created in a2billing it displayed

"request timeout" somewhere i missed the configuration, please help me to
resolve this error .

Thanks,

Gokul.
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Re: [asterisk-users] restricting sip users to a certain useragent

[A J Stiles]

On Tuesday 29 Jun 2010, Tarek Sawah wrote:
> . is it possible to
> force the agents (users) to use a certain UserAgent which is the one
> built-in our system?  this way will prevent the agents we are restricting
> them to only be able to dial through the software which is already
> restricted to their seats in the call center.. but someone might sniff
> around .. and get the sip username and password assigned to him and use it
> through Zoiper or any other softphone to make calls .

If someone is *that* determined, what will stop them from modifying the 
user-agent string in some Open Source softphone?

-- 
AJS

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[asterisk-users] How to Add IP address to SIP Domain

[DHAVAL INDRODIYA]

Dear All,

I have Asterisk and Kamailio Configuration.

everything works fine, now the situation is like i have following Dial
pattern in Dialplan.

exten => s,n, Dial(SIP/1...@glbvoice.com,20,m)

now in my /etc/hosts i have following entry

192.168.1.30 glbvoice.com

then call get forwarded to kamailio and everything is working fine

now question is if i want add one more domain like abc.com so for that i
need to add every entry in /etc/hosts file.

is there anyway to resolve it out, Means if SIP wants to send each call to
192.168.1.30 , but without entry in /etc/hosts.


regards
Dhaval.
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Re: [asterisk-users] transfering active call to user's voicemail

[Gordon Henderson]

On Tue, 29 Jun 2010, Rustam Kovhaev wrote:

> Hi there,
>
> I would like to setup up my Asterisk to do this:
> receptionist answers the call, caller says he wants to leave a
> voicemail message for Ashleigh, receptionist transfers the call to
> Ashleigh's voicemail
>
> I guess It has something to do with dynamic features, or probably
> blind transfer to special ext. might do it
>
> what would you recommend?

If Ashleigh is extnsion 321, then define a 2nd extension 321* which calls 
Voicemail(321,us) then reception can simply do a blind transfer to 321*

Gordon

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