date:20110305

Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread BryantZ



On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com wrote:

 Hi
 
 I have two Asterisk Server:
 
 The first server A, all phone are connected
 The Second server B only route call to a lot of SIP supplier
 
 the server A sent:
 
 ; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten = _X.,3,Hangup
 
 
 anyone know if it's possible to add the CDR Accountcode to this process
 for get it on the second server B ?
 
 i want the same accountcode on the 2 servers
 
 thanks
 Olivier
 
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Send the account code as a custom header variable encode it on A and read it on 
B. You can send any variables you want using this method. I currently send 
about 10 variables on switch transfers. If you need an example ping me back and 
I will send one when I get in the office.

Bryant

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Re: [asterisk-users] Help Asterisk / API / Perl

2011-03-05 Thread Захаров Антон


Hello!

Try to use ${CHANNEL} instead of agi_type.

It will be like this:

$typ = $AGI-get_variable('CHANNEL');
@tmp_array=split(/\//, $typ);
$typ = $tmp_array[0];

and

$src=$AGI-get_variable('cdr(src)');

On 05.03.2011 10:25, Olivier CALVANO wrote:

Hi

i want use the API on my asterisk 1.6, but i have a small problems :

In extension, i start it :
 exten =  _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems

but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)

but that's don't work:

use Asterisk::AGI;
use lib /var/lib/asterisk/agi-bin;
$AGI = new Asterisk::AGI;
$typ = $AGI-get_variable('agi_type');

$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');

anyone know this problems ?

thanks
Olivier

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Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread BryantZ


On Mar 5, 2011, at 8:52 AM, brya...@zktech.com wrote:

 
 
 On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com wrote:
 
 Hi
 
 I have two Asterisk Server:
 
 The first server A, all phone are connected
 The Second server B only route call to a lot of SIP supplier
 
 the server A sent:
 
 ; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
   exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
   exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
   exten = _X.,3,Hangup
 
 
 anyone know if it's possible to add the CDR Accountcode to this process
 for get it on the second server B ?
 
 i want the same accountcode on the 2 servers
 
 thanks
 Olivier
 
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 Send the account code as a custom header variable encode it on A and read it 
 on B. You can send any variables you want using this method. I currently send 
 about 10 variables on switch transfers. If you need an example ping me back 
 and I will send one when I get in the office.
 

Just noticed you are using IAX I don't think my method works with IAX. That is 
why I use SIP between systems. Someone correct me if there is a way to send 
custom variables with IAX.

Bryant

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[asterisk-users] Prepaid Billing other than A2Billing

2011-03-05 Thread bilal ghayyad

Hi All;

Any one advise for open source prepaid billing other than A2Billing that can 
work with Asterisk and it is rich by features (for large business)?

Regards
Bilal


  

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Re: [asterisk-users] Help Asterisk / API / Perl

2011-03-05 Thread Steve Edwards


On Sat, 5 Mar 2011, Olivier CALVANO wrote:


i want use the API on my asterisk 1.6, but i have a small problems :

$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');


'agi_type' is part of the AGI environment, not a channel variable.

Read the documentation for your AGI library to see how to access the AGI 
environment variables -- the cruft Asterisk writes to the STDIN of your 
AGI before any of your requests.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread Steve Edwards


On Sat, 5 Mar 2011, brya...@zktech.com wrote:

Send the account code as a custom header variable encode it on A and 
read it on B. You can send any variables you want using this method. I 
currently send about 10 variables on switch transfers. If you need an 
example ping me back and I will send one when I get in the office.


Just noticed you are using IAX I don't think my method works with IAX. 
That is why I use SIP between systems. Someone correct me if there is a 
way to send custom variables with IAX.


You can pass cruft between Asterisk servers via IAX using the caller ID 
name.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Configuration for Multiple PRI cards

2011-03-05 Thread Elliot Murdock

Hello All,

How does one go about creating a dahdi configuration file for multiple
PRI cards?

Thanks,
Elliot

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Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread Tilghman Lesher

On Sat, Mar 5, 2011 at 11:52 AM, Steve Edwards
asterisk@sedwards.com wrote:
 On Sat, 5 Mar 2011, brya...@zktech.com wrote:

 Send the account code as a custom header variable encode it on A and read
 it on B. You can send any variables you want using this method. I currently
 send about 10 variables on switch transfers. If you need an example ping me
 back and I will send one when I get in the office.

 Just noticed you are using IAX I don't think my method works with IAX.
 That is why I use SIP between systems. Someone correct me if there is a way
 to send custom variables with IAX.

 You can pass cruft between Asterisk servers via IAX using the caller ID
 name.

In 1.6.2 and above, you can set arbitrary variables with IAXVAR() on
one side and retrieve
them on the other side.

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Re: [asterisk-users] Gosub and 'h' (again?)

2011-03-05 Thread Faisal Hanif

Well a solution for you to put original context name in variable and then
use that variable in goto statement on h.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Friday, March 04, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)

Problem as follows:

[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)

[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)

This works fine if the caller listens to all the 'tt-monkeys' and let's the
system hangup.  You get the hang up in the 'default' context.

But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context.  This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).

Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?

I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby the
Gosub can empty the return stack and return to the original place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].

But what does that mean in English ;)?

Thanks

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Re: [asterisk-users] GXW4004 - lines get stuck

2011-03-05 Thread Faisal Hanif

1-Check signaling type on gateway PSTN ports

2-Set RTP timeout in SIP trunk.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, March 04, 2011 7:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] GXW4004 - lines get stuck

Hi,

I have an issue with a GWX4004 used a as a VoIP trunk to PSTN lines
converter.  In some instances, lines get stuck (both parties hang up, but
the GXW4004 status shows off hook for the lines). It stays like this until
reboot.

Is there a specific setting I should be looking for? I couldn't find
anything about that specifically.

Mike

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Re: [asterisk-users] Help Asterisk / API / Perl

2011-03-05 Thread Faisal Hanif

AstPP  jbilling

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, March 05, 2011 10:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help Asterisk / API / Perl

On Sat, 5 Mar 2011, Olivier CALVANO wrote:

 i want use the API on my asterisk 1.6, but i have a small problems :

 $typ don't have SIP or IAX, same test without succes:
 $typ = $AGI-get_variable('type');

'agi_type' is part of the AGI environment, not a channel variable.

Read the documentation for your AGI library to see how to access the AGI
environment variables -- the cruft Asterisk writes to the STDIN of your AGI
before any of your requests.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Early codec selection / negotiation

2011-03-05 Thread Francois Marier

Hi,

This seems to be a fairly common question, but I have Googled for this quite
a bit and looked at the Asterisk documentation/book and haven't been able to
find an answer.

My question is:

  Can I get my IP phone to select a different codec depending on the final
  destination of each call?

I've got these things connected to my Asterisk box:

- Snom 300 phone   (supports g729  and ulaw)
- PSTN Peer(supports g729  and ulaw)
- Remote Asterisk Peer (supports speex and ulaw)

Currently, it's configured like this:

  [snom300]
  disallow=all
  allow=ulaw
  
  [pstnpeer]
  disallow=all
  allow=ulaw
  
  [asteriskpeer]
  disallow=all
  allow=speex

which translates to this:

  Snom300 ---ulaw--- (pass-thru) ---ulaw PSTNPeer
  Snom300 ---ulaw--- (transcode) ---speex--- AsteriskPeer

In other words, my Snom phone always talks to my Asterisk box using the ulaw
codec. My Asterisk box then makes PSTN calls using ulaw and Asterisk calls
using speex (transcoding in the case of speex).

What I'd like to get is this:

  (1) Snom300 ---g729--- (pass-thru) ---g729 PSTNPeer
  (2) Snom300 ---ulaw--- (transcode) ---speex--- AsteriskPeer

I can get (1) by using this config:

  [snom300]
  disallow=all
  allow=g729 ; only allow g729
  
  [pstnpeer]
  disallow=all
  allow=g729

and I can get (2) by using this config:

  [snom300]
  disallow=all
  allow=ulaw ; only allow ulaw
  
  [asteriskpeer]
  disallow=all
  allow=speex

but I can't get both of them to work at the same time since the Snom phone
always connects to my Asterisk box using its prefferred codec.

If I configure the phone like this:

  [snom300]
  disallow=all
  allow=g729 ; preferred codec
  allow=ulaw

then (2) will fail because it's trying to do this:

  Snom300 ---g729--- (transcode) ---speex--- AsteriskPeer

and it can't transcode g729 to speex without a patent license.

If I configure the phone like this:

  [snom300]
  disallow=all
  allow=ulaw ; preferred codec
  allow=g729

then (1) will fail because it's trying to do this:

  Snom300 ---ulaw--- (transcode) ---g729 PSTNPeer

This is the best description of the problem I've found online:

  
http://fonality.com/trixbox/forums/trixbox-forums/open-discussion/codec-selection-negotiation-and-tweaking

but unfortunately it doesn't come with a solution.

Is there a way to prevent my IP phone from always connecting to my Asterisk
box using its preferred codec or is that simply impossible?

Cheers,
Francois

-- 
Francois Marier identi.ca/fmarier
http://feeding.cloud.geek.nz  twitter.com/fmarier

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[asterisk-users] ignore this test

2011-03-05 Thread sean darcy


I can't seem to send anything. Let's see if this shows up.


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[asterisk-users] imsdroid on droidX to asterisk: No matching peer found

2011-03-05 Thread sean darcy


sip.conf:

[imsdroid]
type=friend
;;auth=md5
;;defaultuser=imsdroid
secret=mysecret
host=dynamic
context=cloud-out
qualify=60
dtmfmode=auto
insecure=port,invite
callerid=IMSDroid client imsdroid
disallow=all
allow=ulaw

I've tried with and without defaultuser and secret.

sip show peer imsdroid:

  * Name   : imsdroid
  Secret   : Set
  MD5Secret: Not set
  Remote Secret: Not set
  Context  : cloud-out
  Subscr.Cont. : Not set
..
 Def. Username: imsdroid
..

On the Droid X, imsdroid:

Public Identity:  imsdroid
Private Identity:  imsdroid
Password: mysecret
Realm:  asterisk

I've also tried identity in form imsdr...@asterisk.ip.addr.

From the imsdroid Quick Start:

Public Identity: The public visible identifier where you are willing to 
receive calls or any demands, either a SIP or tel URI (e.g. 
tel:+3310 or sip:b...@open-ims.test).


For a basic SIP client, this should be a SIP URI (a.k.a SIP address).

Private Identity: A unique identifier assigned to a user (or UE) by the 
home network. It could be either a SIP URI (e.g. sip:b...@open-ims.test), 
a tel URI (e.g. tel:+3310) or any alphanumeric string (e.g. 
b...@open-ims.test or bob).


For a basic SIP client, the IMPI should coincide with their 
authentication name.


Password: Your password.

Realm: The realm is the name of the domain to authenticate to. It should 
be a valid SIP URI (e.g. sip:open-ims.test).


sip debug:


--- SIP read from UDP:my.ip.addr:34778 ---
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP my.ip.addr:34778;branch=z9hG4bK741407665;rport
From: sip:imsdroid;tag=629978444
To: sip:imsdroid
Contact: 
sip:(null)@my.ip.addr:34778;transport=udp;expires=60;+g.oma.sip-im;language=en,fr;+g.3gpp.smsip;+g.oma.sip-im.large-message;audio;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs;+g.3gpp.cs-voice
Call-ID: 3e130372-8de8-a6b4-6250-36307973a88f
CSeq: 2010551627 REGISTER
ontent-Length: 0
Max-Forwards: 70
Authorization: Digest 
username=imsdroid,realm=asterisk,nonce=39f59329,uri=sip:asterisk,response=2ba48ca360d592ca183ba6706e6feae9,algorithm=MD5
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
Privacy: none
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
User-Agent: IM-client/OMA1.0 IMSDroid/v1.2.366 (doubango r550)
P-Preferred-Identity: sip:imsdroid
Supported: path

-
--- (16 headers 0 lines) ---
Sending to my.ip.addr:34778 (no NAT)

--- Transmitting (no NAT) to my.ip.addr:34778 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
my.ip.addr:34778;branch=z9hG4bK741407665;received=my.ip.addr;rport=34778
From: sip:imsdroid;tag=629978444
To: sip:imsdroid
Call-ID: 3e130372-8de8-a6b4-6250-36307973a88f
CSeq: 2010551627 REGISTER
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0




--- Transmitting (no NAT) to my.ip.addr:34778 ---
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 
my.ip.addr:34778;branch=z9hG4bK741407665;received=my.ip.addr;rport=34778
From: sip:imsdroid;tag=629978444
To: sip:imsdroid;tag=as44c80a4c
Call-ID: 3e130372-8de8-a6b4-6250-36307973a88f
CSeq: 2010551627 REGISTER
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0



[Mar  6 03:38:28] NOTICE[7794]: chan_sip.c:23511 handle_request_register: Registration from 
'sip:imsdroid' failed for 'my.ip.addr:34778' - No matching peer found



I must be missing something glaringly obvious. Any help appreciated.

sean


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Re: [asterisk-users] Prepaid Billing other than A2Billing

2011-03-05 Thread Pezhman Lali

I think a2billing is the best billing opensource system, but try astbill,
new url http://astbss.org/

http://astbss.org/but if you want to setup a large system select
enterprise system, these systems are useful for small and med networks.
best

On Sat, Mar 5, 2011 at 8:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi All;

 Any one advise for open source prepaid billing other than A2Billing that
 can work with Asterisk and it is rich by features (for large business)?

 Regards
 Bilal




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[asterisk-users] fail2ban + asterisk

2011-03-05 Thread Pezhman Lali

Dear
this note is only for fresh administrators don't think about asterisk
security.
I found fail2ban very useful for anti asterisk hacking, so I want to share
it with fresh admins.
some hackers try your sip or iax2 ip with a lot of username/password, may be
after 1 million try, one username/password was accepted.  so in 2-3 hours,
they use all of the credit of the hacked user.
fail2ban, runs as service, and checks the logs, and blocks the suspicious
IPs.

for more info:

http://www.fail2ban.org/wiki/index.php/Asterisk
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk

best
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