date:20100701

On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
I've never used it (I'm a 1.2 Luddite), but I would be very interested in 
anything that looks like a real language for writing dialplans.

That's why I'm interested in using Lua to write dialplan scripts,
besides the fact that due to its size, Lua is a good solution for
embedded Asterisk appliances.

Not much feedback on this feature. I guess the fact that it's part of
1.6 means that few people gave it a try yet.


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Re: [asterisk-users] How to stop intruder from registering sip?

On Wed, Jun 30, 2010 at 11:50:49PM -0500, Tilghman Lesher wrote:
 On Wednesday 30 June 2010 18:38:51 Steve Edwards wrote:
  On Sun, 13 Jun 2010, Tilghman Lesher wrote:
   I would generally suggest something a little more deterministic (where
   101 is your extension):
  
   $ echo '101This is a salt' | sha1sum
   22c3c098bfc2289396af84ecfb1ab77419a6537e
 
  Aside from being 8 characters longer, why do you prefer sha1sum to md5sum?
 
 The use of MD5 is gradually being displaced, as crypto attacks are getting
 better.  Since SHA1 is usually the replacement, I went with it, since it's
 also likely to be available on systems.  While SHA1 will eventually succumb to
 the same attacks as MD5, due to its larger bitstrength, it has quite a few
 years left in it, before we need to start thinking about SHA256 or SHA512 to
 replace it.

So, assuming I can relatively easily come up with another phrase that
gives the same md5sum as the one of '101This is a salt', what does it
help me with breaking the next extension?

I prefer shorter names. An md5 checksum is too long as-is. Maybe simply
get the first 8 characters from it and hope they are unique. For a small
sample size (I suspect even a few 1000-s here would be small enough) I
would not expect any collisions.

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Re: [asterisk-users] p2p or p2mp for BRI

On Thu, Jul 01, 2010 at 01:21:23PM +0530, pranav jawale wrote:
 Hello,
 
 I'm a graduate student. We are setting up an IVR system for research purpose
 on a BRI channel. (We can't afford PRI line as its cost is about 10x of the
 BRI). The line will be connected to the CTI card. Using asterisk server we
 will be recording the calls.
 
 I'm confused about whether we should apply for point-to-point connectivity
 or point-to-multipoint. P2P costs more than P2MP and
 we do not intend to invest money if not needed. I would like to ask your
 advice on whether for our purpose we need p2p or would p2mp suffice?

The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
protocol, and PtMP is an extension of its logic to make ISDN (BRI)
phones behave somewhat like analog phones: allow you to connect several
of them on the same line.

I suspect that this is really the last thing you need. It makes things
more complicated and less determenistic. However I suspect that if you
ask the phone company for a PTP connection, they'll assume you are a
business customer (connecting his own PBX, rather than multiple
phones). If you can get PTP without paying extra - use it.

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

2010-07-01 Thread Doug Lytle

CunningPike wrote:
 On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com  wrote:


 We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great.



There is a much newer patch for 1.4 that can be found at:

https://issues.asterisk.org/view.php?id=8824

But, it won't apply cleanly on the latest 1.4 series.  It's like 4 
versions back.  Once I get into work, I'll post the version I'm running 
it on.

Doug



-- 
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Re: [asterisk-users] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1

On Wed, Jun 30, 2010 at 05:56:27PM -0500, Alex Villacís Lasso wrote:

 I have reproduced this stream of warnings on another machine with 
 asterisk-1.4.33.1 and dahdi-2.3.0.1, and also with other card types 
 (OpenVox with 1 E1 port, Sangoma with 2 T1 ports, Rhino with 2 T1 
 ports), so I do not think the particular driver is an issue. The 
 question I have is this: is this warning message something to be 
 expected from ports with RED alarms? Or is this message a symptom of a 
 deeper misconfiguration? Since I am the package manager for the Elastix 
 project (http://www.elastix.org), I am the one who can solve 
 misconfigurations, if any.

This pointless warning appears in upstream as well (both in 1.6.2 and
after the rewrite in trunk).

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Re: [asterisk-users] p2p or p2mp for BRI

2010-07-01 Thread pranav jawale

Thank you for your replies.

@Olivier

Yes. The telco here (in Mumbai, India) charges more for p2p. See their
tariff http://mtnlmumbai.in/telecomservices/isdntariff.html#bratariff
It is mentioned in Charges for point to point connectivity : ISDN BRA
Lines that extra charges would be applied for p2p. They have not mentioned
any charges for p2mp so I'm assuming this is what I will be getting p2mp by
default.

We plan to do speech recognition of the collected data, therefore would not
prefer data passed over simulated channel.

@Tzafrir
Yes p2mp might complicate the thing. It seems I have to explicitly ask them
for P2P while applying. I mean that if I do not explicitly ask them for p2p,
will
my card work (for whichever default connection they provide for ISDN
handset) ?

On Thu, Jul 1, 2010 at 3:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:

 On Thu, Jul 01, 2010 at 01:21:23PM +0530, pranav jawale wrote:
  Hello,
 
  I'm a graduate student. We are setting up an IVR system for research
purpose
  on a BRI channel. (We can't afford PRI line as its cost is about 10x of
the
  BRI). The line will be connected to the CTI card. Using asterisk server
we
  will be recording the calls.
 
  I'm confused about whether we should apply for point-to-point
connectivity
  or point-to-multipoint. P2P costs more than P2MP and
  we do not intend to invest money if not needed. I would like to ask your
  advice on whether for our purpose we need p2p or would p2mp suffice?

 The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
 protocol, and PtMP is an extension of its logic to make ISDN (BRI)
 phones behave somewhat like analog phones: allow you to connect several
 of them on the same line.

 I suspect that this is really the last thing you need. It makes things
 more complicated and less determenistic. However I suspect that if you
 ask the phone company for a PTP connection, they'll assume you are a
 business customer (connecting his own PBX, rather than multiple
 phones). If you can get PTP without paying extra - use it.

 --
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 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +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] Pbx_lua vs. calling lua thru AGI?

2010-07-01 Thread Faisal Hanif


 Hi,

I am in process of merging all my AGIs+Dialplan to a single LUA 
dialplan. It seems much interesting to me spacial LUA tables which allow 
me to support a complete object like programming. Yet I did not 
completed / tested.



Regards,

*Faisal Hanif*On 7/1/2010 2:37 PM, Gilles wrote:

On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
asterisk@sedwards.com  wrote:

I've never used it (I'm a 1.2 Luddite), but I would be very interested in
anything that looks like a real language for writing dialplans.

That's why I'm interested in using Lua to write dialplan scripts,
besides the fact that due to its size, Lua is a good solution for
embedded Asterisk appliances.

Not much feedback on this feature. I guess the fact that it's part of
1.6 means that few people gave it a try yet.


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Re: [asterisk-users] Pbx_lua vs. calling lua thru AGI?

On Thu, 01 Jul 2010 15:22:33 +0500, Faisal Hanif fai...@vopium.com
wrote:
I am in process of merging all my AGIs+Dialplan to a single LUA 
dialplan. It seems much interesting to me spacial LUA tables which allow 
me to support a complete object like programming. Yet I did not 
completed / tested.

Thanks for the input. I'm interested in any information you might
have.


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Re: [asterisk-users] p2p or p2mp for BRI

2010-07-01 Thread Philipp von Klitzing

Hi!

 The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
 protocol, and PtMP is an extension of its logic to make ISDN (BRI) phones
 behave somewhat like analog phones: allow you to connect several of them
 on the same line.

In other words:

While you *must* have exactly one central PBX with PTP, you *can* have a 
central PBX or you can use BRI decentrally because PtMP is organized as a 
bus system. In the case of HylaFax, for example, this means that with 
PtMP you can run chan_capi (Asterisk) and capi4hylafax in parallel on the 
line so that they do not depend on each other at all (no need for 
IAXmodem, no fax timing issues).

Numbering: With PTP you can usually obtain a number block of 2-10 
consecutive numbers, while with PtMP you will probably get 2 or 3 (up to 
10) individual numbers that do not have to be consecutive. At least 
that's the way it works in Germany and Belgium.

PTP can be easily enlarged with more BRI lines, whereas with PtMP this is 
more difficult (due to line hunting from Telco to you, choosing an 
outgoing DID/MSN etc).

Finally: Make sure that ISDN equipment (card, gateway) and driver 
supports the mode that you decide for. In the case of Asterisk chan_capi 
is a good choice, mISDN is not - but if you are in the US then I am have 
no cluee how good National ISDN support is with chan_capi.

Philipp


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Re: [asterisk-users] Pbx_lua vs. calling lua thru AGI?

On Thu, 01 Jul 2010 01:32:08 +0200, Gilles codecompl...@free.fr
wrote:
I'm taking a look at how to write scripts to be called from the
dialplan, and saw pbx_lua mentioned.

I'm not having much luck adding the pbx_lua module to Asterisk (on a
Ubuntu 10.04) :-/

# apt-get install lua5.1 liblua5.1-0 liblua5.1-0-dev

# cd /usr/src/asterisk/asterisk
# make menuconfig

PBX Modules  XXX pbx_lua : Depends on: lua(E)

Does someone know what other packages are required to pbx_lua to be
available as an option?

Thank you.


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[asterisk-users] Originate multiple channels

2010-07-01 Thread Deepesh D

Hello,

Is it possible to use the asterisk manager interface to originate
multiple channels?

like
Action: Originate
Channel: SIP/101SIP/102

So that both extensions 101 and 102 rings simultaneously.

I am using asterisk manager interface over http.

Thanks

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

On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle supp...@drdos.info wrote:
 CunningPike wrote:
 On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com  wrote:


 We use the patch in https://issues.asterisk.org/view.php?id=6643. Works 
 great.



 There is a much newer patch for 1.4 that can be found at:

 https://issues.asterisk.org/view.php?id=8824

 But, it won't apply cleanly on the latest 1.4 series.  It's like 4
 versions back.  Once I get into work, I'll post the version I'm running
 it on.

 Doug



This is the version that went into trunk for 1.8. It should send the
remote party id without dialplan changes. I had looked into using it
with 1.6.1 and 1.6.2. However due to the number of changes since the
patch was merged I was worried that I would introduce bugs. The
previous patch is simple, but does require a one line dial plan
change.

On the previous patch I posted for 1.6.2 I also have a 1.6.1 version.
It compiles but hasn't been tested. Let me see if I can quickly put
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.

Ryan

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Re: [asterisk-users] Pbx_lua vs. calling lua thru AGI?

On Thu, Jul 01, 2010 at 01:21:31PM +0200, Gilles wrote:
 On Thu, 01 Jul 2010 01:32:08 +0200, Gilles codecompl...@free.fr
 wrote:
 I'm taking a look at how to write scripts to be called from the
 dialplan, and saw pbx_lua mentioned.
 
 I'm not having much luck adding the pbx_lua module to Asterisk (on a
 Ubuntu 10.04) :-/
 
 # apt-get install lua5.1 liblua5.1-0 liblua5.1-0-dev
 
 # cd /usr/src/asterisk/asterisk

Re-run ./configure

 # make menuconfig
 
 PBX Modules  XXX pbx_lua : Depends on: lua(E)
 
 Does someone know what other packages are required to pbx_lua to be
 available as an option?

The Debian asterisk package depends on liblua5.1-0-dev and builds
pbx_lua just fine.

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Re: [asterisk-users] Originate multiple channels

2010-07-01 Thread Paul Belanger

On Thu, Jul 1, 2010 at 7:49 AM, Deepesh D deep.d2...@gmail.com wrote:
 So that both extensions 101 and 102 rings simultaneously.

Yes, or use a local channel to dial multiple extensions.

-- 
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Re: [asterisk-users] Originate multiple channels

2010-07-01 Thread Zeeshan Zakaria

Unfortunately not. I did it a few times using a php script using a 'which'
loop to create multiple call files. You can also do it in a dialplan which
is a slow process. I have it described at:

http://ilovetovoip.com/2010/03/calling-multiple-extensions-and-let-them-all-answer/

Zeeshan A Zakaria

--
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On 2010-07-01 8:18 AM, Deepesh D deep.d2...@gmail.com wrote:

Hello,

Is it possible to use the asterisk manager interface to originate
multiple channels?

like
Action: Originate
Channel: SIP/101SIP/102

So that both extensions 101 and 102 rings simultaneously.

I am using asterisk manager interface over http.

Thanks

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

2010-07-01 Thread Doug Lytle

Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



Thank you!

Doug

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Re: [asterisk-users] How to stop intruder from registering sip?

2010-07-01 Thread William Stillwell (Lists)



Also, technically your 101This is a salt is stronger than your SHA1 Hash.

Let's say you stick with the 17 character password

You are using 0-9, a-z, A-Z, and space.

0-9 = 10
a-z = 26
A-Z = 26
Space = 1
Total Possible Values = 63

17^63 = 3.2982384238829760312713680399948e+77

Your sha1 is using 0-9, a-f

0-9 = 10
a-f = 6

40^16 = 4294967296

Your best defense would be:

1) don't use the extension # as the username
2) don't use any form of word out of any dictionary for user or password
3) try to make username/password as long as possible

4) don't use the [default] in the extension.conf (just in case you missed
something, and someone gets in somewhere.

5) use fail2ban or some other type of system to block ip's of remote systems
that attempt to authenticate more then 5 times in a minute and fail. (less,
whatever your feel is sufficient)




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, July 01, 2010 5:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to stop intruder from registering sip?

On Wed, Jun 30, 2010 at 11:50:49PM -0500, Tilghman Lesher wrote:
 On Wednesday 30 June 2010 18:38:51 Steve Edwards wrote:
  On Sun, 13 Jun 2010, Tilghman Lesher wrote:
   I would generally suggest something a little more deterministic (where
   101 is your extension):
  
   $ echo '101This is a salt' | sha1sum
   22c3c098bfc2289396af84ecfb1ab77419a6537e
 
  Aside from being 8 characters longer, why do you prefer sha1sum to
md5sum?
 
 The use of MD5 is gradually being displaced, as crypto attacks are getting
 better.  Since SHA1 is usually the replacement, I went with it, since it's
 also likely to be available on systems.  While SHA1 will eventually
succumb to
 the same attacks as MD5, due to its larger bitstrength, it has quite a few
 years left in it, before we need to start thinking about SHA256 or SHA512
to
 replace it.

So, assuming I can relatively easily come up with another phrase that
gives the same md5sum as the one of '101This is a salt', what does it
help me with breaking the next extension?

I prefer shorter names. An md5 checksum is too long as-is. Maybe simply
get the first 8 characters from it and hope they are unique. For a small
sample size (I suspect even a few 1000-s here would be small enough) I
would not expect any collisions.

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[asterisk-users] mISDN install on Asterisk 1.6 failing

2010-07-01 Thread Shaun Wingrin

Hi,

Has anyone had experience installing it?
yum install asterisk-chan_misdn 
I'ts the latest Trixbox Distro version and same issues exists if add in the 
Trixbox repo.
FAILS as per below:
I have a ISDN single port PCI BRI card installed and detected.
__
Loaded plugins: fastestmirror, kmod
Loading mirror speeds from cached hostfile
 * addons: www.ftp.saix.net
 * base: www.ftp.saix.net
 * extras: www.ftp.saix.net
 * updates: www.ftp.saix.net
Excluding Packages from CentOS-5 - Addons
Finished
Excluding Packages from CentOS-5 - Base
Finished
Excluding Packages from CentOS-5 - Extras
Finished
Excluding Packages from CentOS-5 - Updates
Finished
Setting up Install Process
Resolving Dependencies
-- Running transaction check
--- Package asterisk-chan_misdn.i386 0:1.4.22-3 set to be updated
-- Processing Dependency: libsuppserv.so.0 for package: asterisk-chan_misdn
-- Processing Dependency: libmISDN.so.0 for package: asterisk-chan_misdn
-- Processing Dependency: libisdnnet.so.0 for package: asterisk-chan_misdn
-- Finished Dependency Resolution
asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems
  -- Missing Dependency: libmISDN.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems
  -- Missing Dependency: libisdnnet.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems
  -- Missing Dependency: libsuppserv.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
Error: Missing Dependency: libisdnnet.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
Error: Missing Dependency: libmISDN.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
Error: Missing Dependency: libsuppserv.so.0 is needed by package 
asterisk-chan_misdn-1.4.22-3.i386 (trixbox)
 You could try using --skip-broken to work around the problem
 You could try running: package-cleanup --problems
package-cleanup --dupes
rpm -Va --nofiles --nodigest
The program package-cleanup is found in the yum-utils package.


Shaun Wingrin
VOIP Telecoms Solution Provider
BSc. (Elec. Eng.) UP

A1 Telecoms cc
Office: 010-590-0222
Mobile: 082-449-6273
Fax: 0880-11-640-5633
Email: sha...@a1telecoms.co.za

Keeping you TALKING for LESS!
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Re: [asterisk-users] Pbx_lua vs. calling lua thru AGI?

On Thu, 1 Jul 2010 15:26:27 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Re-run ./configure

Ah, hadn't thought of this :-/

The Debian asterisk package depends on liblua5.1-0-dev and builds
pbx_lua just fine.

Yes, it did compile after re-running ./configure, make menuconfig,
make.

I'll check how to use extension.lua instead of extensions.conf, and
see how it goes.

Out of curiosity, what are the added-value of pbx_lua + extensions.lua
over calling the Lua interpreter through AGI?

Thank you.


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[asterisk-users] AppDial in CEL Data

2010-07-01 Thread Nic Colledge

Hi,

I am using CEL to more accurate billing information with some success. However 
there is an ambiguity in the CEL data when multiple destinations are specified 
in the DIAL command.

For example, if I have 
Dial(SIP/outboundA/100SIP/outboundA/101SIP/outboundB/200SIP/outboundB/201) 
this is reflected in the dial command data that shows up in CEL.

The problem is in some situations it is difficult to tell which one of these 
destinations answered the call because the CEL_Answer event does not store the 
destination number anywhere. It would be nice if the appdata of the CEL_Answer  
event were the part of the dial command which was used to create that channel 
so say SIP/outboundA/101 rather than (Outgoing Line).

I am currently assuming the order in which the channels were created 
corresponds to the order in which the destinations appear in the dial command 
to find the answered destination. This works fine most of the time, it only 
fails when one of more of the outgoing channels could not be created.

Would it be possible to change this?

Thanks,
Nic

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[asterisk-users] Brute force attacks

2010-07-01 Thread Ishfaq Malik


Hi

We've just noticed attempts (close to 20 attempts, sequential peer 
numbers) at guessing peers on 2 of out servers and thought I'd share the 
originating IPs with the list in case anyone wants to firewall them as 
we have done


109.170.106.59
112.142.55.18
124.157.161.67

Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062
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Re: [asterisk-users] call file question

2010-07-01 Thread Jeff LaCoursiere


On Wed, 30 Jun 2010, Steve Edwards wrote:

 Now I whipped up a C program to create a call file to do the same thing
 from the command line:

 [snip]
 fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);

 I don't see exten *71 in custom-callfwd.

Doh!  That was the problem.  In FreePBX I made *71 the feature code to 
access that context, and it was still in my head when I made the callfile.


 Why are you using a local channel in your call file?


That was the meat of the question, actually.  I want to create a single 
leg with a callfile - just the outbound call.  All other times I have used 
callfiles I was creating two legs and bridging them.  Is there a better 
way to do what I am attempting?


  fprintf(callfile, Application: Playback\n);
  fprintf(callfile, Data: hello-world\n);
 [snip]

 When I run this it creates the call file and I see this in the console:

 -- Attempting call on Local/*...@custom-callfwd/n for application
 Playback(hello-world) (Retry 1)

 What does the call file look like before you mv it to the spool directory?


Exactly the above fprintf lines...

Thanks,

j

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Re: [asterisk-users] mISDN install on Asterisk 1.6 failing

2010-07-01 Thread Philipp von Klitzing

Hi!

 Has anyone had experience installing it? 
 yum install asterisk-chan_misdn 
 I'ts the latest Trixbox Distro version and same issues exists if add in
 the Trixbox repo. FAILS as per below

Please search this list for recent messages on mISDN, or Google it. 

You will find that mISDN v1 does not work with current kernels anymore, 
and that you would need mISDN v2, vor which howerver this (or was when I 
last checked) there is no chan_misdn available for Asterisk so that you 
would have to resort to use LCR which kind of defeats the purpose when 
run next Asterisk.  

mISDN v1 is not reliable and will give you all sorts of trouble. Look at 
other solutions like CAPI and chan_capi, mISDN v2, external BRI gateway, 
...

Philipp


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[asterisk-users] Remote Party ID issue



Hi,
 
i have the same problem. Trying to use the dialplan function CONNECTEDLINE() 
this way
 
 Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)})
 Set(CONNECTEDLINE(num)=${EXTEN})

ends with
 
[Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function 
CONNECTEDLINE not registered

Same happens trying function CALLEDID.
 
I am using Asterisk 1.6.1.20.
 
What do i have to do to use this function or alternatively the function 
CALLEDID() described in bug 8824?
 
Thanks in advance for help.


 
Ondrej Valousek wrote:
 Hello,
 
 Did anyone manage to force asterisk to put Remote-party-ID attribute on 
 the SIP outgoing call? I.e. When A calls B, I want that A gets a name of  B 
 displayed on his phone.
 Note that name of A gets displayed on the B's phone fine, but this is 
 not what I want.
 This works with Cisco Call manager fine - the RPID is sent as a part of 
 the response to the SIP INVITE this way:
 
 
 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport
 From: Ondrej Valousek sip:7775 at 192.168.60.20 sip:7775 at 
 192.168.60.20 ;tag=as4786d518
 To: sip:1098 at 192.168.62.12 sip:1098 at 192.168.62.12 
 ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104
 Date: Tue, 30 Mar 2010 13:53:15 GMT
 Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20
 CSeq: 102 INVITE
 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
 SUBSCRIBE, NOTIFY
 Allow-Events: presence
 *Remote-Party-ID: Paul Ryan sip:1098 at 192.168.62.12 sip:1098 at 
 192.168.62.12 ;party=called;screen=yes;privacy=off*
 Contact: sip:1098 at 192.168.62.12:5060 sip:1098 at 192.168.62.12:5060 
 Content-Length: 0
 
 
 But I can not make it working with Asterisk. Does anyone have any glue 
 how to achieve this WITHOUT patching asterisk? I am happy to upgrade to 
 the latest/greatest version, I just do not want to patch.
 Many thanks,
 
 Ondrej
 

This feature is in Asterisk trunk and will be present in the upcoming 1.8 
release. By setting sendrpid=yes on A's phone, Asterisk will send a 
Remote-Party-ID header that corresponds to what Asterisk received from B. Also, 
there is a CONNECTEDLINE() dialplan function that can be used to send this 
information prior to a call. I actually gave a presentation on this topic at 
Astricon last year, but for some reason the Astricon '09 archive does not seem 
to have my presentation video available.

Mark Michelson


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

2010-07-01 Thread bruce bruce

Thanks a lot. I will look into it.

On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wcse...@selbytech.comwrote:

 On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce bruceb...@gmail.com wrote:

 Thanks a lot.

 -Bruce


 On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote:

 Hi bruce,

 SIPDefault.conf


 I think you need one of the newer XML config files for the 7965.  I have an
 example that works with a 7941 on my website (you can find the link my
 signature), I think with a little adaptation you can make it work with a
 7965.


 --
 Thanks,
 --Warren Selby
 http://www.selbytech.com

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Re: [asterisk-users] Problem in establish call from a2billing users.

2010-07-01 Thread bruce bruce

Yes, you are missing a whole bunch of configurations from creating SIP users
to making sure they show as peers on Asterisk to making sure you use dnid,
etc.You probably might want to search google for some configuration help

On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan alagudr...@gmail.comwrote:

 Hi All,

 I installed a2billing with asterisk FreePBX  .  I can able to login and
 make a call with FreePBX but

 when i am using the users which is created in a2billing the call was not
 established . I know somewhere i missed

 the configuration please any one help me to resolve this issue . Thanks in
 advance.

 regards,

 gokul.,

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Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Steve Howes

On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:

 [Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function 
 CONNECTEDLINE not registered
 Same happens trying function CALLEDID.
  
 I am using Asterisk 1.6.1.20.
  
 What do i have to do to use this function or alternatively the function 
 CALLEDID() described in bug 8824?
 

Isn't CONNECTEDLINE only in trunk?
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Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.

In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.

exten = 
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

Ryan


asterisk-1.6.1.20-called-rpid.patch
Description: Binary data


asterisk-1.4.33.1-called-rpid.patch
Description: Binary data
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Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Andrew Latham

Only in trunk...(1.8)

~
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 Thu, Jul 1, 2010 at 11:02 AM, Steve Howes steve-li...@geekinter.net wrote:
 On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:

 [Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function 
 CONNECTEDLINE not registered
 Same happens trying function CALLEDID.

 I am using Asterisk 1.6.1.20.

 What do i have to do to use this function or alternatively the function 
 CALLEDID() described in bug 8824?


 Isn't CONNECTEDLINE only in trunk?
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Re: [asterisk-users] Remote Party ID issue

Sorry, what does this mean? Only in trunk?





-Original Message-
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:02 pm
Subject: Re: [asterisk-users] Remote Party ID issue


On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:
 [Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function 
ONNECTEDLINE not registered
 Same happens trying function CALLEDID.
  
 I am using Asterisk 1.6.1.20.
  
 What do i have to do to use this function or alternatively the function 
ALLEDID() described in bug 8824?
 
Isn't CONNECTEDLINE only in trunk?
- 

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Re: [asterisk-users] Brute force attacks

2010-07-01 Thread John Timms

On Thu, Jul 1, 2010 at 9:16 AM, Ishfaq Malik i...@pack-net.co.uk wrote:

  Hi

 We've just noticed attempts (close to 20 attempts, sequential peer
 numbers) at guessing peers on 2 of out servers and thought I'd share the
 originating IPs with the list in case anyone wants to firewall them as we
 have done

 109.170.106.59
 112.142.55.18
 124.157.161.67

 Ish
 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd

 Office:   0161 660 3062

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We have noticed the same sort of activity on our server. The originating IP
addresses attempting access were:

204.9.204.145 (hosted at U.S. Colo, I believe)
91.203.132.149 (Nephax)
130.70.157.186 (University of Louisiana)
61.160.121.46 (Chinanet)
109.170.0.10 (ReasonUP Ltd)

--
John Timms
IT Department - Gnoso Inc.
j...@gnoso.com
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Re: [asterisk-users] Update the LCD with the callee's name after dialing


Sounds great.

Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.

Thanks.






-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after 
dialing


On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --

The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
ompile but need to be tested to verify that they work. I have the
.6.2.9 version in production and plan to put the 1.6.1.20 version in
ometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
emote party id with something like below. Otherwise check out the
revious FreePBX 2.7 patch.
exten = 
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan

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

On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.

Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.

Thanks.

-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing

On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:

together one for 1.4 that compiles. I'll post both to the list
hopefully later today.

Thank you!

Doug

The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.

In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.

exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

Ryan

If you installed Asterisk from source you just need to patch and
recompile / install.

cd asterisk-version
patch -p1 ../asterisk-verson-called-rpid.patch
make install

Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.

Ryan

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Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Andrew Latham

http://svnview.digium.com/svn/asterisk/trunk/


~
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 Thu, Jul 1, 2010 at 11:25 AM,  unsero...@aol.com wrote:
 Sorry, what does this mean? Only in trunk?



 -Original Message-
 From: Steve Howes steve-li...@geekinter.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:02 pm
 Subject: Re: [asterisk-users] Remote Party ID issue

 On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:

 [Jul  1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function
 CONNECTEDLINE not registered
 Same happens trying function CALLEDID.

 I am using Asterisk 1.6.1.20.

 What do i have to do to use this function or alternatively the function
 CALLEDID() described in bug 8824?


 Isn't CONNECTEDLINE only in trunk?
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Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Steve Howes


On 1 Jul 2010, at 16:25, unsero...@aol.com wrote:

 Sorry, what does this mean? Only in trunk?

If you look in the post you quoted

This feature is in Asterisk trunk and will be present in the upcoming 1.8 
release.

First sentence.

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


Thanks a lot.

Applying the patch gives me a

Hunk #5 failed at 9881






-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after 
dialing


On Thu, Jul 1, 2010 at 11:29 AM,  unsero...@aol.com wrote:
 Sounds great.

 Could you please give me a hint how to install the patch?
 Sorry for my stupid question but I'm a newbie to Asterisk.

 Thanks.



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:06 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


 The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
 compile but need to be tested to verify that they work. I have the
 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
 sometime this weekend.

 In you are just using Asterisk in the dialplan you can set the called
 remote party id with something like below. Otherwise check out the
 previous FreePBX 2.7 patch.

 exten =
 
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

 Ryan
If you installed Asterisk from source you just need to patch and
ecompile / install.
cd asterisk-version
atch -p1  ../asterisk-verson-called-rpid.patch
ake install
Otherwise if your using trixbox, etc you would probably want to grab
heir SRPMS, add the patch to the spec file, and rebuild them. However
hat is outside of the scope of this mailing list.
Ryan
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Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Steve Howes

On 1 Jul 2010, at 16:56, unsero...@aol.com wrote:
 
 Sorry, i wanted to know what is in trunk means.
 So it seems to mean is in the pipeline for the next version.

DON'T reply to people off list. And stop bloody top posting.

Steve

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

On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote:
Thanks a lot.

Applying the patch gives me a

Hunk #5 failed at 9881

-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing

On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.

Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.

Thanks.

On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:

together one for 1.4 that compiles. I'll post both to the list
hopefully later today.

Thank you!

Doug

In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.

exten =

100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

Ryan

If you installed Asterisk from source you just need to patch and
recompile / install.

cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install

Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.

Ryan

Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.

Ryan

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Re: [asterisk-users] How to stop intruder from registering sip?

2010-07-01 Thread Tilghman Lesher

On Thursday 01 July 2010 07:43:38 William Stillwell (Lists) wrote:
 Also, technically your 101This is a salt is stronger than your SHA1 Hash.

 Let's say you stick with the 17 character password

 You are using 0-9, a-z, A-Z, and space.

 0-9 = 10
 a-z = 26
 A-Z = 26
 Space = 1
 Total Possible Values = 63

 17^63 = 3.2982384238829760312713680399948e+77

 Your sha1 is using 0-9, a-f

 0-9 = 10
 a-f = 6

 40^16 = 4294967296

That would only be true if you used random characters in your 17-character
passphrase.  In fact, English text has somewhere between 0.6 and 1.5 bits of
randomness per letter, whereas an SHA1sum has no more than 4 bits of
randomness per letter.  Let's assume the higher number of randomness for
your English text, which gives us 1.5 * 17, which is 25.5 bits of randomness.
Note that the prefix 3 characters have ZERO randomness per character, as they
are deterministic from the extension.  That gives an even less 21 bits of
randomness.  SHA1 cryptographic sums have no more than 160 bits of randomness.

I say no more than, because, given knowledge of the algorithm used to
determine passwords, the sum is reduced to the number of bits of randomness in
the source material.  You cannot generate randomness by applying a
deterministic algorithm.  However, given that the source material for the hash
sum is of a smaller bit strength than the comparative strength of the hash
algorithm, your difficulty of guessing the password is not reduced any by
using the hash algorithm for generative purposes.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Dial SIP channel with no registration, timeout before CONGESTION?

2010-07-01 Thread Jack Bates

What determines how long SIP channel waits, when you dial a peer with no
registration, before returning ${DIALSTATUS} CONGESTION?

When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?

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Re: [asterisk-users] How to stop intruder from registering sip?

2010-07-01 Thread Mark Deneen

On Thu, Jul 1, 2010 at 12:53 PM, Tilghman Lesher tles...@digium.com wrote:


 That would only be true if you used random characters in your 17-character
 passphrase.  In fact, English text has somewhere between 0.6 and 1.5 bits
 of
 randomness per letter, whereas an SHA1sum has no more than 4 bits of
 randomness per letter.  Let's assume the higher number of randomness for
 your English text, which gives us 1.5 * 17, which is 25.5 bits of
 randomness.
 Note that the prefix 3 characters have ZERO randomness per character, as
 they
 are deterministic from the extension.  That gives an even less 21 bits of
 randomness.  SHA1 cryptographic sums have no more than 160 bits of
 randomness.

 I say no more than, because, given knowledge of the algorithm used to
 determine passwords, the sum is reduced to the number of bits of randomness
 in
 the source material.  You cannot generate randomness by applying a
 deterministic algorithm.  However, given that the source material for the
 hash
 sum is of a smaller bit strength than the comparative strength of the hash
 algorithm, your difficulty of guessing the password is not reduced any by
 using the hash algorithm for generative purposes.



With this in mind, I'll be sure to forge my passwords from Chinese text from
now on.
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Re: [asterisk-users] How to stop intruder from registering sip?

2010-07-01 Thread Dave Platt

 That would only be true if you used random characters in your 17-character
 passphrase.  In fact, English text has somewhere between 0.6 and 1.5 bits of
 randomness per letter, whereas an SHA1sum has no more than 4 bits of
 randomness per letter.  Let's assume the higher number of randomness for
 your English text, which gives us 1.5 * 17, which is 25.5 bits of randomness.
 Note that the prefix 3 characters have ZERO randomness per character, as they
 are deterministic from the extension.  That gives an even less 21 bits of
 randomness.  SHA1 cryptographic sums have no more than 160 bits of randomness.
 
 I say no more than, because, given knowledge of the algorithm used to
 determine passwords, the sum is reduced to the number of bits of randomness in
 the source material.  You cannot generate randomness by applying a
 deterministic algorithm.  However, given that the source material for the hash
 sum is of a smaller bit strength than the comparative strength of the hash
 algorithm, your difficulty of guessing the password is not reduced any by
 using the hash algorithm for generative purposes.

Agreed, on all points.

Any deterministic method of this sort (e.g. hashing together the
extension name with a constant-per-site salt) is vulnerable to a
brute-force guessing attack against the salt.  If the person who
set it up used a ordinary, easily-remembered phrase as the salt,
then the security of *all* of the secrets is tied to the guessability
of this phrase.  Brute-force dictionary attacks against plain-language
words and phrases have been quite successful in the past... I've heard
it said that on any multi-user system having more than a handful of
users, the odds of one of those users having a guessable password
are often 50% or better.

I'm not in favor of using this sort of deterministic scheme
(e.g. HASH(salt + public info)) for determining per-station
secrets, no matter which hash algorithm is used.  Instead, I
recommend the scheme I originally proposed - use a random-
number generator (or a cryptographically-string pseudorandom
generator, fed with some entropy from an external unpredictable
source) to generate individual secrets.  I make three arguments:

-  The resulting secrets (i.e. strings of hexadecimal digits)
   are equally hard, or equally easy, for the end-users to deal with
   (assuming that we're talking about equal numbers of digits).  Neither
   scheme has an advantage here.

-  Once set up, both systems are equally easy to use and administer...
   press a button and generate a secret.

-  The random- or pseudo-random method produces secrets which don't
   depend at all on the extension numbers (or user names, or other
   public information), are independent from one another, and are
   essentially immune to dictionary and other guessing attacks.  The
   only way to break them is via a full brute-force search... and
   successfully finding one extension's secret by brute-force search
   doesn't help you at all in finding any other extension's.  Assuming
   a good random-number generator, the amount of entropy (randomness)
   in the secrets is essentially equal to (2 ^ number-of-bits).

   None of these things is true of a deterministic-hashing scheme...
   if the salt can be guessed or determined, *every* extension's secret
   has been broken, and you have to immediately change *every* configuration
   in order to secure your system.  Salts based on dictionary words and
   phrases have far less randomness in them than their length would
   imply, and that means that the resulting secrets are less random...
   generating longer secret strings doesn't fix this, and can simply
   give a false sense of security.




-

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[asterisk-users] rename External Directory

2010-07-01 Thread Brad Zynda

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1



Hey Guys,

Saw your article at www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx
and had a question regarding the directory on a 7960 POS3-08-6
not running call manager.

I quickly figured out each directory only holds 32 spots and need to implement
an A-M and N-Z but the phone labels the directories as External Directory

I also realized changing the Titleabc/Title or the Promptabc/Prompt
does nothing...

Is there a way to change the External Directory to CompanyName A-M etc...?

Thanks,

Brad
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (Darwin)

iEYEARECAAYFAkws5SIACgkQhzJ5NSeNtkheSACcD2FFN3eIV0+sfZnBrJWrN10l
3v8An0IoJHVh/2chw+bcn4Q0WZuveoJ/
=LhNY
-END PGP SIGNATURE-

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Re: [asterisk-users] Problem with extensions in IVR and queues

2010-07-01 Thread Anahi Ludueña


Hi, we've just been able to find the problem. Apparently it was related to the 
softphone. We've installed another one and the call is performed ok.
Thanks!






Anahi Ludueña
 



From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 19:59:14 +
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues








Ups, sorry, that CLI output is related to my other problem (the options of IVR 
doesn't responde when the call is from landline or cell phone).
I'll put the correct CLI output...
Thanks,





Anahi Ludueña
 



From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 19:50:00 +
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues








This is the CLI output, the dialplan is the one that the Elastix creates when 
somebody sets the followme... I don't know what part you want I post here...
Thanks,

-- Executing [4...@from-internal:1] GotoIf(SIP/9050-001185aa, 
0?ext-local|4010|1) in new stack
-- Executing [4...@from-internal:2] Macro(SIP/9050-001185aa, 
user-callerid|) in new stack
-- Executing [...@macro-user-callerid:1] Set(SIP/9050-001185aa, 
AMPUSER=9050) in new stack
-- Executing [...@macro-user-callerid:2] GotoIf(SIP/9050-001185aa, 
0?report) in new stack
-- Executing [...@macro-user-callerid:3] ExecIf(SIP/9050-001185aa, 
1|Set|REALCALLERIDNUM=9050) in new stack
-- Executing [...@macro-user-callerid:4] Set(SIP/9050-001185aa, 
AMPUSER=9050) in new stack
-- Executing [...@macro-user-callerid:5] Set(SIP/9050-001185aa, 
AMPUSERCIDNAME=CALLPBX) in new stack
-- Executing [...@macro-user-callerid:6] GotoIf(SIP/9050-001185aa, 
0?report) in new stack
-- Executing [...@macro-user-callerid:7] Set(SIP/9050-001185aa, 
AMPUSERCID=9050) in new stack
-- Executing [...@macro-user-callerid:8] Set(SIP/9050-001185aa, 
CALLERID(all)=CALLPBX 9050) in new stack
-- Executing [...@macro-user-callerid:9] ExecIf(SIP/9050-001185aa, 
0|Set|CHANNEL(language)=) in new stack
-- Executing [...@macro-user-callerid:10] GotoIf(SIP/9050-001185aa, 
0?continue) in new stack
-- Executing [...@macro-user-callerid:11] Set(SIP/9050-001185aa, 
__TTL=64) in new stack
-- Executing [...@macro-user-callerid:12] GotoIf(SIP/9050-001185aa, 
1?continue) in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [...@macro-user-callerid:19] NoOp(SIP/9050-001185aa, Using 
CallerID CALLPBX 9050) in new stack
-- Executing [4...@from-internal:3] GotoIf(SIP/9050-001185aa, 1?skipdb) 
in new stack
-- Goto (from-internal,4010,5)
-- Executing [4...@from-internal:5] Set(SIP/9050-001185aa, __NODEST=) 
in new stack
-- Executing [4...@from-internal:6] Set(SIP/9050-001185aa, 
__BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa) in new stack
-- Executing [4...@from-internal:7] Set(SIP/9050-001185aa, 
__BLKVM_BASE=4010) in new stack
-- Executing [4...@from-internal:8] Set(SIP/9050-001185aa, 
DB(BLKVM/4010/SIP/9050-001185aa)=TRUE) in new stack
-- Executing [4...@from-internal:9] Set(SIP/9050-001185aa, RRNODEST=) 
in new stack
-- Executing [4...@from-internal:10] Set(SIP/9050-001185aa, 
__NODEST=4010) in new stack
-- Executing [4...@from-internal:11] Set(SIP/9050-001185aa, 
RecordMethod=Group) in new stack
-- Executing [4...@from-internal:12] Macro(SIP/9050-001185aa, 
record-enable|4010|Group) in new stack
-- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa, 
1?check) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa, 
recordingcheck|20100630-154030|1277926830.37214) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:5] MacroExit(SIP/9050-001185aa, ) 
in new stack
-- Executing [4...@from-internal:13] Set(SIP/9050-001185aa, 
RingGroupMethod=ringallv2) in new stack
-- Executing [4...@from-internal:14] Set(SIP/9050-001185aa, 
_FMGRP=4010) in new stack
-- Executing [4...@from-internal:15] GotoIf(SIP/9050-001185aa, 
0?doconfirm) in new stack
-- Executing [4...@from-internal:16] Macro(SIP/9050-001185aa, 
dial|20|tr|4010) in new stack
-- Executing [...@macro-dial:1] GotoIf(SIP/9050-001185aa, 1?dial) in 
new stack
-- Goto (macro-dial,s,3)
-- Executing [...@macro-dial:3] AGI(SIP/9050-001185aa, dialparties.agi) 
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'CALLPBX' number is '9050'
  dialparties.agi: USE_CONFIRMATION:  'FALSE'
  dialparties.agi: RINGGROUP_INDEX:   ''
  dialparties.agi: Methodology of ring

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