[asterisk-users] Cdr Logs modify Disposition on Unsuccessful call

2012-03-26 Thread Vinod Dharashive 
Hi Team,

I would like capture SS7 Error Code in CDRs, Specifically of outbound call
from the asterisk. calls generated using .call file.

In extension.conf extens gets excuted on successful call only , So that on
h extension reason of hangup is captured. But i am not aware of any
provision that capture on Unsuccessful call.

please guide on this or suggest any patch.

Thanks
Vinod d
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Re: [asterisk-users] T.38 troubles

2012-03-26 Thread Jean-Denis Girard 
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Hi Steve,

Le 26/03/2012 14:50, Steve Underwood a écrit :
> Your log shows the Mediatrix GW has problems. It sends a DCS signal to
> the Asterisk box, but doesn't following it with TCF as it should. The
> asterisk box times out waiting for TCF and tries to take recovery action
> which fails.

Thanks for your analysis. Could it be a configuration problem on the
Mediatrix?

> Spandsp has some workarounds for bugs in Mediatrix boxes. They usually
> work OK.

What would you suggest then, is there anything to do to enable the
workarounds? spandsp-0.0.6pre20 is the latest available, or should I try
spandsp-20120324? As far as I know, there is no firmware update for the
Mediatrix...


Thanks,
- -- 
Jean-Denis Girard

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http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
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Re: [asterisk-users] T.38 troubles

2012-03-26 Thread Steve Underwood 

Hi Jean-Denis,

Your log shows the Mediatrix GW has problems. It sends a DCS signal to 
the Asterisk box, but doesn't following it with TCF as it should. The 
asterisk box times out waiting for TCF and tries to take recovery action 
which fails.


Spandsp has some workarounds for bugs in Mediatrix boxes. They usually 
work OK.


Regards,
Steve


On 03/27/2012 08:02 AM, Jean-Denis Girard wrote:

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Hi list,

I'm having difficulties when receiving faxes from the PSTN with this
relatively simple installation:
PSTN<--PRI-->  GW<--T.38-->  Asterisk

The gateway is a Mediatrix 3301 (firmware Dgw 2.0.14.251). It's
configured to transmit faxes as T.38. I may have missed something in its
configuration, but it does switch to T.38 when a fax is detected. On the
Asterisk side, I'm using 10.2.1 with spandsp-0.0.6-pre20 and ReceiveFax
from res_fax_spandsp. ${FAXSTATUS} returns FAILED and ${FAXERROR}
"Disconnected after permitted retries".

I did a network capture, attached to this mail: from my understanding,
T.38 is accepted by Asterisk, then there seems to be some UDPTL traffic,
which I don't understand...

Why does it fail, and what is wrong? I'd appreciate if someone could
send me advice / suggestions.


Thanks,
- - --
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
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[asterisk-users] T.38 troubles

2012-03-26 Thread Jean-Denis Girard 
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Hash: SHA1

- -BEGIN PGP SIGNED MESSAGE-
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Hi list,

I'm having difficulties when receiving faxes from the PSTN with this
relatively simple installation:
   PSTN <--PRI--> GW <--T.38--> Asterisk

The gateway is a Mediatrix 3301 (firmware Dgw 2.0.14.251). It's
configured to transmit faxes as T.38. I may have missed something in its
configuration, but it does switch to T.38 when a fax is detected. On the
Asterisk side, I'm using 10.2.1 with spandsp-0.0.6-pre20 and ReceiveFax
from res_fax_spandsp. ${FAXSTATUS} returns FAILED and ${FAXERROR}
"Disconnected after permitted retries".

I did a network capture, attached to this mail: from my understanding,
T.38 is accepted by Asterisk, then there seems to be some UDPTL traffic,
which I don't understand...

Why does it fail, and what is wrong? I'd appreciate if someone could
send me advice / suggestions.


Thanks,
- - --
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
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fax-ast10.pcap.bz2
Description: BZip2 compressed data
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[asterisk-users] Collaboration Call Center Integrated with Asterisk "web and email"

2012-03-26 Thread bilal ghayyad 
Hi All;

Is there a collaboration contact center (hope to be open source) Integrated 
with Asterisk (hope with vicidial), so the agent will be able to receive chat 
or emails sessions and deal with the customer. If the agent in a call with the 
customer, then he will not get chat session. Is there like this software?

Regards
Bilal

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Re: [asterisk-users] Authentication: username and password, also to be from the LAN

2012-03-26 Thread Danny Nicholas 
IMO it is an "additional restriction" not an either/or.  If you do something
like allowing guest logins, it is probably a good practice to limit the
range they can login from.  Even for your known users, it isn't a bad idea
if your box is open to the internet in general.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Monday, March 26, 2012 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Authentication: username and password, also to
be from the LAN

Is that now permit and deny are used for. To specify the acceptable IP
address(es) the user can connect from?
--
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Mar 26, 2012, at 10:11 AM, bilal ghayyad wrote:

> Hi All;
> 
> Is it possible to restrict the authentication to be based on the username
and password and to be allowed for IPs within the LAN (for example,
192.168.10.x)? 
> 
> I do not need it to be based on the IP only and do not need it to be based
on the username and password only, but I need it to be based on the username
& password and to be from the specific range, so if the IP address of the
client was of the range 192.168.10.x then it is allowede to register with
its username and password. No need to specify the IP. 
> 
> If it possible, then is it possible to be a configuration per user?
> 
> Regards
> Bilal
> 
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Re: [asterisk-users] Authentication: username and password, also to be from the LAN

2012-03-26 Thread Jim Dickenson 
Is that now permit and deny are used for. To specify the acceptable IP 
address(es) the user can connect from?
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Mar 26, 2012, at 10:11 AM, bilal ghayyad wrote:

> Hi All;
> 
> Is it possible to restrict the authentication to be based on the username and 
> password and to be allowed for IPs within the LAN (for example, 
> 192.168.10.x)? 
> 
> I do not need it to be based on the IP only and do not need it to be based on 
> the username and password only, but I need it to be based on the username & 
> password and to be from the specific range, so if the IP address of the 
> client was of the range 192.168.10.x then it is allowede to register with its 
> username and password. No need to specify the IP. 
> 
> If it possible, then is it possible to be a configuration per user?
> 
> Regards
> Bilal
> 
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] Authentication: username and password, also to be from the LAN

2012-03-26 Thread bilal ghayyad 
Hi All;

Is it possible to restrict the authentication to be based on the username and 
password and to be allowed for IPs within the LAN (for example, 192.168.10.x)? 

I do not need it to be based on the IP only and do not need it to be based on 
the username and password only, but I need it to be based on the username & 
password and to be from the specific range, so if the IP address of the client 
was of the range 192.168.10.x then it is allowede to register with its username 
and password. No need to specify the IP. 

If it possible, then is it possible to be a configuration per user?

Regards
Bilal

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Re: [asterisk-users] Settings problems of Asterisk as client

2012-03-26 Thread Leandro Dardini 
Your problem originate from the use of insecure option. Using this option,
the peer is authenticated using the registration ip and not the user and
password.

Leandro
Il giorno 26/mar/2012 05:48, "YeungJoe"  ha scritto:

>  Hello All,
>
> I am Asterisk user, and right now I have some troubles about Asterisk As
> Client settings.
>
> Here are my envrionments:
>
> Asterisk-1.8.5.0
>
> ---
> Server Settings(IP:172.16.70.121)
>
> extensions.conf
>
>
> [from-internal-200]
> exten => _X.,1,Dial(SIP/${EXTEN})
> exten => _X.,n,Hangup()
>
> end of extensions.conf/
>
>
> sip.conf///
> [101]
> type=friend
> username=101
> secret=101
> host=dynamic
> allow=all
> context=from-internal-101
>
>
> [102]
> type=friend
> username=102
> secret=102
> host=dynamic
> allow=all
> context=from-internal-102
>
>
> [200]
> type=friend
> username=200
> secret=200
> host=dynamic
> allow=all
> context=from-internal-200
> end of sip.conf///
>
> ---
> Client Settings(IP:172.16.70.124:
>
> //extensions.conf//
> [from-sip-101]
> exten => s,1,Noop(SIP-101)
>
> [from-sip-102]
> exten => s,1,Noop(SIP-102)
> end of extensions.conf/
>
>
> /sip.conf//
> [general]
> register => 101:101@172.16.70.121
> register => 102:102@172.16.70.121
>
> [101]
> type=peer
> username=101
> secret=101
> insecure=invite,port
> host=172.16.70.121
> context=from-sip-101
>
> [102]
> type=peer
> username=102
> secret=102
> insecure=invite,port
> host=172.16.70.121
> context=from-sip-102
> //end of sip.conf/
> ---
>
> Right now, I am able to register extensions 101 and 102 to
> server(172.16.70.121).
> and I can dial from SIP extension 200 to 101 or 102, if I dial 101, it
> will be
> routed to 101, and 101 is ringing. This is OK. but if I dial 102, it also
> be routed 101, I don't know why, because
> according to my SIP knowledges it should be routed to 102 as they are
> different contexts.
>
> BTW, Client peer is also based on Asterisk.
>
> I am a newbie of SIP, if you need more info I will provide.
> Please help! Thanks!
>
>
> Joe.Yeung
> ***
> *
>
>
>
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