Re: [asterisk-users] asterisk for small home phone system

2012-10-25 Thread Roger Burton West
On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote:
 - Is the Linksys SPA3102 a good piece of hardware for this type of setup or 
 is there something cheaper?  Perhaps a card that can go right into the Linux 
 box?

I'm using an OpenVox A400 (with an FXO module), which Asterisk can
drive directly.

 - Would we configure our SIP clients on our iphones to login directly to 
 Asterisk running on my home Linux box?  I have 18MB/2.5MB internet service 
 with a static IP so this wouldn't be a problem.

That would be the simplest approach (modulo firewalls). If you already
have another SIP provider, you could configure your home asterisk to
forward calls to that...

R

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Re: [asterisk-users] asterisk for small home phone system

2012-10-25 Thread Roger Burton West
On Thu, Oct 25, 2012 at 11:33:06AM -0700, Matthew Hixson wrote:
Is there any reason a regular old voicemodem wouldn't work?

IME the voice quality and reliability are pretty grotty. If you find
one that works, great!

R

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Re: [asterisk-users] How to stop ringing when incoming PSTN call is answered externally?

2012-05-23 Thread Roger Burton West
On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote:
The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone) the asterisk
extensions continue to ring one more time.  Is there a way to have
Asterisk drop an incoming PSTN call as soon as it's answered?

I have the same problem, and earlier discussion here suggests it's
insoluble.

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Re: [asterisk-users] using Wifi smartphones as SIP clients

2012-05-07 Thread Roger Burton West
On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote:
What about phones like the Unidata WPU-7800 (
http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
experience with those? Would these also suffer from connection
losses?

I've been using a UTStarcom GF-210 for the last year and more as my
personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP
is slightly better than 2G, getting it to talk to Asterisk is no problem
at all, but certainly if you're moving from one wifi device to another
you will get dropped calls. If that's your use case, it's going to be
that way whatever hardware you use - I haven't seen any implementations
of 802.11F or 802.11r in the field.

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Re: [asterisk-users] using Wifi smartphones as SIP clients

2012-05-07 Thread Roger Burton West
On Mon, May 07, 2012 at 09:14:36PM +0200, Hans Witvliet wrote:

Hope that these are better that the utstar F1000:
Keep on re-chargibg as battery is empty in no-time, and security is
lousy; just  wep, no wpa.

WPA and WPA2. Battery lasts about a day in dual mode, much longer in
2G-only of course. And at UKP30 they may be worth a punt even if you end
up upgrading to something else.


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Re: [asterisk-users] Asterisk 1.8.12.0-rc1

2012-04-12 Thread Roger Burton West
On Thu, Apr 12, 2012 at 01:14:25PM -0700, motty.cruz wrote:

Can this be acomplish? I hope I explained better. 

Yes, no problem.

First, get the two servers talking to each other (I like IAX for this,
but SIP also works). If NAT is a concern, there are various ways round
it (I like VPN tunnels).

Then set up the dialplan on the public server to route the call to the
other machine.


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[asterisk-users] MessageSend, SIP, and call files

2012-04-10 Thread Roger Burton West
As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.

After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and something like this almost works:

Channel: Local/8902
Application: MessageSend
Set: MESSAGE(body)=messagebody
Data: sip:glowworm
Data: sip:glowworm

but (a) I need that reserved local number to let the call work at all
(the number just does an Answer(), Wait(10), Hangup) and (b) I can't
seem to set the sender's name. That ought to be the second Data
parameter; actually the second one seems to determine where the message
goes, and whatever I set the first one to the sender name always comes
up as asterisk. (Specifically, in the packet capture, I have

From: asterisk sip:asterisk@[ipaddr]:5060

.) Now, I _can_ achieve the desired result, but only by having _another_
local number that does

exten = 8901,n,SET(MESSAGE(body)=${msg_out_body})
exten = 8901,n,MessageSend(${msg_out_to},${msg_out_from})

and setting up the callfile with:

Extension: 8901
Set: msg_out_to=glowworm
Set: msg_out_from=sip:FROM@DOMAIN

at which point the message will appear to originate from FROM (note that
if I put a display name component in the msg_out_from it gets ignored -
but that is the terminals' peculiarity). But that's ugly. Has anyone got
this working with a relatively straight callfile setup?

While I'm writing, does Asterisk 10 have any way to send a SIP message
that isn't text/plain?

Roger

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Re: [asterisk-users] MessageSend, SIP, and call files

2012-04-10 Thread Roger Burton West
On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote:
This is what core show applications in 10.1.3 shows
 SendDTMF: Sends arbitrary DTMF digits
   SendFAX: Sends a specified TIFF/F file as a FAX.
 SendImage: Sends an image file.
  SendText: Send a Text Message.
   SendURL: Send a URL.
You are using sendtext - you might want to use sendurl instead.

Those are all about sending data in an existing channel, though -
the trick is that I don't _have_ a channel, which is presumably why
MessageSend exists. Is there a way to set up a channel without ringing
the phone?


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Re: [asterisk-users] Standard UIDs, especially for asterisk?

2011-11-15 Thread Roger Burton West
On Tue, Nov 15, 2011 at 04:42:05PM +, Tony Mountifield wrote:
But it sounds like it is distro-specific.

No, it's system-specific. Debian for example will assign UIDs out of the
relevant range based on the order in which packages are installed.

Just use the textual UID/GID values, not the numeric ones.

Roger

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Re: [asterisk-users] Asterisk to asterisk IAX trunk

2011-10-11 Thread Roger Burton West
On Tue, Oct 11, 2011 at 02:53:26PM +0100, Jonathan Archer wrote:

How can I get the 5 to stay where it is so that lookups work correctly?
is it part of the outbound CID?

My trunking (prefix 9 to get trunk access from either side of the link)
includes things like:

exten = _9NX.,1,Set(CALLERID(num)=9${CALLERID(num)})
exten = _9NX.,n,Dial(IAX2/remoteserver/${EXTEN:1},,wW)

R

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

2011-09-12 Thread Roger Burton West
On Mon, Sep 12, 2011 at 12:21:06PM -0600, linux guy wrote:

FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have
graphical tools.

To add to what everyone else has said: if you _really_ need to run a
graphical tool on the server, you can always ssh -X into it without
having to have a full desktop installed there.

(As for wireshark: tcpdump on site, then bring the capture file home to
analyse with wireshark. Works for me...)

Roger

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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing

2011-08-12 Thread Roger Burton West
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:

shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')

Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?


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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing

2011-08-12 Thread Roger Burton West
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
Yes, same server, same filesystem...

I don't do Python, but a web search for shutil.move suggests that it
doesn't reliably use the rename syscall. Might be worth shelling out
to your system's mv command.

R

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Re: [asterisk-users] Answering machine answers after pickup a phone.

2011-08-05 Thread Roger Burton West
On Fri, Aug 05, 2011 at 10:59:03AM +0200, Jorge Barreiro wrote:

What I try to do is that, when there is an incoming call from the ouside, if 
someone answers on a phone, then the PBX won't answer.

I have a couple of VoIP phones fed through Asterisk, as well as analogue
phones linked directly to the line. In this case, picking up the
analogue phone stops the VoIP phones ringing (after ten seconds or so).
I don't know whether this would be achievable with the Asterisk
console and soundcard drivers...

Roger

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Re: [asterisk-users] Re : Direct RTP with Asterisk

2011-06-19 Thread Roger Burton West
On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote:
No, I can't, because, it's a different NAT. I try to simulate P2P with 
asterisk.
What you suggest to me ?

I like VPN tunnels. They give you a flat network topology and decent
security.


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Re: [asterisk-users] click to call

2011-06-17 Thread Roger Burton West
On Fri, Jun 17, 2011 at 05:20:39PM +, salaheddine elharit wrote:
i want to use sip 223 in order to call phone number

Is that meant to be the originator or the destination?

Channel: gets the originator; Extension: gets the destination.

Roger

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Re: [asterisk-users] Audio dropping

2011-05-28 Thread Roger Burton West
On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote:

What could the reason be audio in 1 direction is dropping? (Normally from
the Asterisk server to the mentioned SIP clients.) No clear information is
in the logs (it is like the call ended normally) and not all calls are
having problem (most not, but it happens to often for us to start using VoIP
more at the moment).

While the most usual problem is packet filtering / NAT, this generally
manifests as no audio at all in one direction, not a drop in mid-call.
But it's possible that one of the intermediate transit providers is
doing something clever. (Disabling ping, as you mention in your later
email, is often a good indicator of a company with insufficient Clue.)

Are you in a position to tunnel the traffic over a VPN or similarly flat
and unfilterable network link? (This might be a good idea anyway.)

Roger

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Re: [asterisk-users] Skype-like dialing from web page

2011-05-17 Thread Roger Burton West
On Tue, May 17, 2011 at 01:30:33PM -0400, Mike wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a web
page?  As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application.  I'd like to use this but on a
normal softphone (Bria, Xlite, other).

Generate a callfile, setting Channel to point to the softphone (e.g.
SIP/Xlitephone) and Extension to point to the number you want to dial.
(You'll need to specify Context too.)

When the callfile is processed, the softphone will ring; when it's
picked up, it will dial the far end.

Roger

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Re: [asterisk-users] Voicemail Configuration

2011-05-09 Thread Roger Burton West
On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote:
However, I want to record what is said during that time and send it
to a third voicemail box once the caller hangs up without having
pressed 1 or 2.

You could use Monitor to record the whole call, then use an AGI to do
something with it on hangup if the other conditions haven't been
satisfied...?

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Re: [asterisk-users] Best Scripting Language

2011-04-01 Thread Roger Burton West
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can anyone suggest which is the best scripting language for Asterisk or any
telecom device?

Depends on the other parameters. Perl is great for rapid development,
but I wouldn't run it per-call on a box taking hundreds of calls per
second. (Ditto Ruby and Python.) C will be much faster, but it's more
effort to write and debug.

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Re: [asterisk-users] Dialplan help: hang up incoming call and call the number back

2011-03-28 Thread Roger Burton West
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:

Is there a better way of handling the post-hangup
processing?

Callfiles?

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Re: [asterisk-users] Detect DTMF tone during call?

2011-02-26 Thread Roger Burton West
On Sat, Feb 26, 2011 at 03:08:02AM -0600, Dan Saul wrote:
I am attempting to create a intercom buzzer system using asterisk as a
back end. Most is figured out except the actual action of buzzing the
door. I need to detect whether a DTMF key was pressed by the the
called party (the resident). Is this possible to do using just a
dialplan? I can't see any options on the Dial command that would lead
to this, am I looking in the wrong place? I looked briefly through the
archive and I heard mentions of AGI, is this what must be used to
accomplish this?

If you want it to be detected within a call, which is what I'd assume,
you'll probably be looking at the applicationmap section within
features.conf.

http://www.voip-info.org/wiki/view/Asterisk+config+features.conf

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Re: [asterisk-users] Unknown calls

2011-02-24 Thread Roger Burton West
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:

Still last night there was a call to a customer. Plz help me figure out the
solution for this problem.

Can you be sure that the call _is_ coming through your Asterisk server,
rather than being the result of random scanning for your customers'
phones?


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[asterisk-users] Carrying context from one server to another?

2011-02-24 Thread Roger Burton West
The relevant part of my setup is something like:

SIP phones - local server - remote server - SIP-to-PSTN provider

I want _some_ of the SIP phones on the local server to be able to get
access to SIP-to-PSTN, but not all of them. The local-to-remote
connection is IAX2 over VPN.

Do I need to set up two separate IAX2 connections, one privileged and
the other not, or can I somehow tag calls from some phones on the local
server so that they're noted as privileged on the remote server?

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Re: [asterisk-users] On-Hold Music

2011-02-15 Thread Roger Burton West
On Fri, Feb 11, 2011 at 04:37:49PM -0600, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a
legal music-on-hold file?

One source of explicitly royalty-free music is the podcasting community:
http://uhort.no/ and http://www.podsafeaudio.com/ both have extensive
libraries.

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Re: [asterisk-users] On-Hold Music

2011-02-15 Thread Roger Burton West
On Tue, Feb 15, 2011 at 08:39:57AM -0600, Danny Nicholas wrote:

Good suggestion, Roger, but this seems like a slippery slope path.
Today's podcaster could be tomorrows ASCAP/BMI member coming back for you?

Doesn't matter if you use music that has been explicitly released as
royalty-free (usually under a CC licence or similar). The URLs I gave
are resources _for_ podcasters; sorry I didn't make that clearer.

R

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Re: [asterisk-users] On-Hold Music

2011-02-15 Thread Roger Burton West
On Tue, Feb 15, 2011 at 09:01:16AM -0600, Danny Nicholas wrote:

Thanks for the tip - got a Norwegian translator for uhort.no?

Anything wrong with
http://translate.google.com/translate?js=nprev=_thl=enie=UTF-8layout=2eotf=1sl=notl=enu=uhort.no
?

R

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Re: [asterisk-users] Using files .call or AMI

2011-02-14 Thread Roger Burton West
On Mon, Feb 14, 2011 at 04:06:10AM +, Edwin Quijada wrote:

How would be the dialplan for this context from-lan ???

This list is for non-commercial support. If you want someone to do the
work for you, I suggest you go elsewhere and offer money.


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Re: [asterisk-users] Using files .call or AMI

2011-02-12 Thread Roger Burton West
On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote:
I have a webpage with information about a customer so in this page the agent 
click a phone number and asterisk do the call and transfer the call to agent 
if this call is answered.

Usually it's the other way round: the agent's phone rings, and when he
picks it up the other end gets dialled. That's trivial with call files:

Channel: (local channel ID for agent)
Context: (context for calling local channel)
Extension: (remote party's phone number)

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Re: [asterisk-users] Using files .call or AMI

2011-02-12 Thread Roger Burton West
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote:
This works for me.! but the agent has to dial the number ?
How could be the context for do this ? U can give an example ?

I'm using this to place calls from local IP-phones over the PSTN. So my
script will generate, say:

Channel: SIP/lanphone
Context: from-lan
Extension: 08001234567

taking the 0800... from the list of customer details.

SIP/lanphone is the ID of the originating phone. Extension is the
sequence the agent would dial if he were placing the call himself.
The originating phone rings; when it's picked up, the Asterisk server
calls the Extension number and bridges the two calls, so the local
agent hears ringing tones from the far end. All the agent has to do is
pick up the phone when it rings and put it down when the call is over.

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Re: [asterisk-users] meetme conference playback of random sound file

2011-02-10 Thread Roger Burton West
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote:

i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself.

Absent an Asterisk-specific solution, how about a separate process which
would link a random file into a fixed pathname? (Fired off from cron,
perhaps.)

Roger

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[asterisk-users] SIP MESSAGE outside calls - state of the art?

2011-02-09 Thread Roger Burton West
I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
SMS messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone:

[Feb  9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received 
message to sip:2...@firedrake.org from Display Name 
sip:mob...@firedrake.org;tag=87739132, dropped it...
  Content-Type:text/plain; charset=UTF-8
  Message: test message

(Packet trace shows a SIP MESSAGE, answered by a 405.)

...and apparently is unable to originate them either; SendText, which
looks as though it ought to be the right way to send them, produces (in
the context of a call, since I can't send the message outside one):

-- Executing [604@default:2] SendText(SIP/mob776-02ba6050, test 
message) in new stack
-- Incoming call: Got SIP response 405 Method Not Allowed back from 
10.0.155.21

even though it's also making a SIP MESSAGE request.

The only documentation I can find talks about a patch and is pretty old:
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging

What I would like is to be able to send a textual message from the phone
into an AGI script (or for other processing), and to return results the
same way. Is anyone doing this with later versions of Asterisk, or
indeed anything else?

Roger

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Re: [asterisk-users] Asterisk on Debian Lenny with timerfd

2011-01-24 Thread Roger Burton West
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock lenny
installation with the updated glibc as well as the latest kernel

At this point the easiest option will be to upgrade to squeeze.

R

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Re: [asterisk-users] Failed SIP registration kicks registered device off?

2011-01-12 Thread Roger Burton West
On Wed, Jan 12, 2011 at 10:13:22AM -0600, Kevin P. Fleming wrote:

His point is valid though... A's registration should not have been
overwritten until B *successfully* registered. A failed attempt to
register should have no effect on the existing registration.

Indeed, the avenue for a brute-force DoS (absent something like
fail2ban) is fairly obvious.


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Re: [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.

2011-01-03 Thread Roger Burton West
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote:

What I want is when a potential client submits his number... the PBX dials the 
number makes an announcement and dials an extension (which is actually a 
cellhopne dahdi member) and makes the connection.

You might try something based on this:

http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out

It's easy to generate a call file which dials the agent's phone, waits
for a pickup, and then dials out.

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Re: [asterisk-users] incoming

2011-01-02 Thread Roger Burton West
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote:
Cool.  So, one Asterisk machine handling up to 100 DID numbers, correct?

As many as you like, modulo memory and CPU requirements.

I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.  Correct?

Depends on how they're presented to you by the DID provider.


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Re: [asterisk-users] How to initiate a two-party call from within Asterisk

2010-12-31 Thread Roger Burton West
On Mon, Nov 29, 2010 at 01:36:17PM -0600, Chris Gentle wrote:

This is click-to-call.  It can be done with the Asterisk Manager Interface
(AMI).  See this site:

Thanks to you and Tilghman for this, though as it turned out it was much
simpler to avoid AMI completely and use the Extension: parameter to an
outgoing call file.

Roger

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Re: [asterisk-users] start services automatically

2010-12-20 Thread Roger Burton West
On Mon, Dec 20, 2010 at 02:34:23PM +, salaheddine elharit wrote:

the 0,1, and 6 are OFF just the 2,3,4,5 are ON ,
and when i reboot the server i found that the service httpd is off  with
command service httpd status and service asterisk status

please advice

This is just one of many problems you will encounter. You need to train
or hire an actual Unix/Linux system administrator.


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Re: [asterisk-users] Specifying DID for outbound calls

2010-12-19 Thread Roger Burton West
On Sun, Dec 19, 2010 at 12:14:11AM -0500, Stephen Reese wrote:

Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf, thanks.

I think I'd probably replace the two outgoing contexts with one, using a
GotoIf to distinguish between the two phones (branching into your
current code).

Alternatively you could give them each a custom context (say phone1 and
phone2); phone1 would include incoming and outgoing1, phone2 would
include incoming and outgoing2.


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Re: [asterisk-users] Call sip:u...@domain.com?

2010-12-17 Thread Roger Burton West
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote:

Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.

How would you _expect_ to be able to specify a destination server from a
telephone keypad?


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Re: [asterisk-users] Configuring server to call SIP numbers on the Net?

2010-12-13 Thread Roger Burton West
On Mon, Dec 13, 2010 at 07:28:58PM +0100, Gilles wrote:
This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix... how should I configure Asterisk so
that SIP clients can dial SIP numbers on the Net, such as those below
to perform an echo test?

If you want to dial a SIP number that's not on your local server, you
need to route it via dialplan logic. You could do this with a prefix
code if you want to be able to dial lots of numbers at the same server:

exten = _9NX.,1,Dial(SIP/user:p...@ekiga.net/${EXTEN:1})

or something more specific if you just want to connect to one:

exten = 602,1,Dial(SIP/user:p...@ekiga.net/*010600)

(Don't quote me on syntax; I don't have any SIP examples handy as I only
use it for local-network calls.)

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Re: [asterisk-users] Asterisk 1.6.2.10 video call

2010-12-06 Thread Roger Burton West
On Mon, Dec 06, 2010 at 03:23:42PM +0100, Jonas Kellens wrote:

I'm trying to set up a video call from my Ekiga client to a
Grandstream GXV3140 IP-phone. The call succeeds but there is no
video.

Try restricting video codec to H.261.

R

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[asterisk-users] How to initiate a two-party call from within Asterisk

2010-11-29 Thread Roger Burton West
The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)

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Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone

2010-11-05 Thread Roger Burton West
On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote:

Anyway, I think the idea of replicating the function into an extension will
work. Any pointers on the best way to accomplish this? I created a new
extension but am unsure what to do next. I thought about the FollowMe
feature but I would have to hardcode the number and I want to be able to
enter a forwarding phone number for the extension using my cell.

You could set up an extension match that triggers on

(feature ID)(access code)(extension)

as it might be, with an access code of 62889:

exten = _*7262889.,1,Set(FWDNUM=${EXTEN:8})

and then put FWDNUM into the astdb or however else you want to handle
it.

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Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100

2010-11-03 Thread Roger Burton West
On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote:

What hardware would I need in the Asterisk so I could hook up some analogue 
extensions? Am I right in thinking I need something like an FXO/FXS card?

Yes, this ought to work. If you're plugging phones into the Samsung it's 
providing an FXS interface, so you'll need an FXO interface to talk to that; if 
you want to connect those analogue phones to Asterisk, you'll also need FXS 
interfaces (though as a short-term fix it would probably be easier to leave 
them plumbed directly into the Samsung box). Getting four modules (each of 
which can be FXO or FXS) on a single card is pretty easy (I use an OpenVox 
A400P from voipon, following recommendations on this list).

You could then connect (some combination of) analogue channels to (some 
combination of) SIP phones, and vice versa to allow outward dialling.

Once you build the Asterisk-only system, you can use the FXO modules to connect 
to analogue PSTN lines (assuming you have a use for this).

Roger


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Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100

2010-11-02 Thread Roger Burton West
On Tue, Nov 02, 2010 at 04:13:01PM +, Ronny Adsetts wrote:

3. Other ways?

It all rather depends on what your proprietary system has been set up to do. 
(If you didn't already have the Samsung box, you wouldn't need to buy one.) 
Dedicated telephony hardware tends to be restricted in all sorts of perverse 
ways to try to make you buy more from the same manufacturer; that'll be your 
biggest problem.

Ideally you would be able to tell your iDCS100 there are multiple VoIP phones 
at this IP address, and connect to the Asterisk server over the LAN. How you 
would go about that, I have no idea; I suspect SIP IP Trunking is what 
Samsung calls this feature. The more work you can  shoft onto the Asterisk 
server, the cleaner this will all be.

In this scenario, the Asterisk server just has a normal network card in it, and 
you shift all your VoIP traffic over the LAN and VPN.



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Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100

2010-11-02 Thread Roger Burton West
On Tue, Nov 02, 2010 at 05:54:27PM +, Ronny Adsetts wrote:

Would it be possible do you know to use the Samsung handsets with an Asterisk 
system? Is it worth even trying to save money here? (I've no idea of the cost 
of VoIP handsets for use with Asterisk).

I've never heard of Samsung handsets in the context of Asterisk, so I'm 
guessing they're Samsung-only. If I were in your shoes I'd go for open 
standards all the way. The cheapest Grandstream SIP phone will run you about 40 
pounds retail and _will_ work with Asterisk - or with anything else that speaks 
SIP.

(And of course with an open platform you can give people softphones on their 
PCs if that's what they prefer - some laptop users do.)

Roger


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Re: [asterisk-users] IAX or SIP - connecting two Asterisk servers together

2010-11-02 Thread Roger Burton West
On Tue, Nov 02, 2010 at 03:20:48PM -0400, Silver Thorne wrote:

I am not looking for someone to do this for me, I am just not really
sure how to get started. Perhaps some suggested reading, examples,
etc?

The simplest approach would be to skip the answering and just dial
through immediately, feeding back the destination's ring tone to the
originator.

Set up an IAX link between the two boxes (you could do it with SIP, but
I found IAX less trouble), then set up an appropriate bit of dialplan
logic on the American box, as it might be:

exten = 4682,1,Dial(IAX2/usern...@eurobox/8873)

Roger

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Re: [asterisk-users] SIP Load Balancing

2010-10-28 Thread Roger Burton West
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.

I think what you need to do here is check/set a variable in the astdb.

(If the variable is 1, set it to 2 and route via A; otherwise, set it to
1 and route via B.)

Translation of this to dialplan logic is left as an exercise for the
student.

R

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Re: [asterisk-users] SIP - no audio behind nat problem

2010-10-15 Thread Roger Burton West
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote:

We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
natted network.

The simplest solution will be to stick another Asterisk box inside the
NAT and tunnel IAX or SIP over a VPN.

R

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Re: [asterisk-users] sound file debug

2010-10-12 Thread Roger Burton West
You have two separate problems here:

(1)

dollars.WAV:  RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit,
mono 8000 Hz

You should have generated this with 16-bit resolution, like all the
others.

(2)

Not sure about the cents - sure it's coming out as 16-bit? Is the file
in the right place?


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[asterisk-users] Asterisk sharing a line with POTS handsets: how to interoperate cleanly?

2010-10-05 Thread Roger Burton West
I now have an OpenVox A400P and it is working well. Thanks to Ade
Vickers for the recommendation, which I second.

However, I need to make a slow transition between a conventional
multiple-extension setup and a full VoIP network on these premises. So
at the moment the Asterisk box shares the PSTN connection with several
conventional analogue handsets.

The desired result for an incoming call is that the Asterisk server will
wait N seconds before answering (which I can arrange easily enough), and
if the call has been answered on one of the handsets by that time the
Asterisk server should ignore it completely. Otherwise it should start
checking CLID, prompting for extensions, and other good stuff, which
again I know how to do.

What is a good approach to making sure the Asterisk server doesn't pick
up a call that has been answered elsewhere? (Ideally in pure dialplan,
but a perl AGI would also do.)

R

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Re: [asterisk-users] minimum card for dahdi timing source ?

2010-10-02 Thread Roger Burton West
On Sat, Oct 02, 2010 at 06:24:24PM +0200, mancyb...@gmail.com wrote:
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock 
timing source for conferences to work properly ?

Can't speak for vicidial, but MeetMe() works fine for me with asterisk
1.4 and ztdummy. I would assume 1.6 with dahdi works similarly...

R

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Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

2010-10-02 Thread Roger Burton West
On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote:
Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to
the PAP2T? do you think the devices comes in with it's external IP rather
than the dyndns domain?

Yes. An IP datagram carries only the source and destination IP
addresses, not the DNS names associated with them. Your firewall _may_
be able to accept a DNS name to block or allow rather than an IP
address, but most don't, and doing so makes you vulnerable to DNS
spoofing attacks.

To go further would be thoroughly off-topic for this list.

Roger

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Re: [asterisk-users] Problems compiling Asterisk on Debian

2010-09-27 Thread Roger Burton West
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:

What should i do?

aptitude install module-assistant
m-a a-i dahdi


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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

2010-09-22 Thread Roger Burton West
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.

Have you considered running Asterisk on Server B as well, and using IAX
to trunk between them? This is working well for me.

Roger

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Re: [asterisk-users] Unable to open vm-INBOXs

2010-09-22 Thread Roger Burton West
On Wed, Sep 22, 2010 at 12:32:21PM +0200, Jonas Kellens wrote:

[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full:
File vm-INBOXs does not exist in any format
[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable
to open vm-INBOXs (format 0x8 (alaw)): No such file or directory

I do not find this particular soundfile on my system.

How are you invoking it? That terminal s on the filename looks rather
unexpected.

R

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Re: [asterisk-users] Bug with Realtime?

2010-09-20 Thread Roger Burton West
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote:
Can we not do pastebin any more?

No, it's just one user with an excessively paranoid and chatty
mailfilter.


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[asterisk-users] Authentication best practice

2010-09-20 Thread Roger Burton West
I am working with a simple follow-me-style service: rather than have
something that rings several phones in turn, the user dials a number (in
the present implementation, unique to that user) to register his
presence at a particular extension.

What's the standard way to protect this from unauthorised use?
Voicemail()-style, where the user has to enter a PIN once the connection
is made? With a very long number, so that number and PIN can be
integrated in the phone's contact list? With a single central number,
where the each user has to enter his own unique identifier _and_ PIN?

Roger

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Re: [asterisk-users] Force ip disconnect after register?

2010-09-13 Thread Roger Burton West
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of 
register attempts.

Consider writing a filter for fail2ban [http://www.fail2ban.org/] that
works on the Asterisk logs?

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Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

2010-09-13 Thread Roger Burton West
On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote:

 [Sep 13 20:14:59] -- Launched AGI Script  
 /var/lib/asterisk/agi-bin/cleanpickup.agi
 [Sep 13 20:14:59]  opruimenpickup.agi: Failed to execute  
 '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied

So check that /var/lib/asterisk/agi-bin/cleanpickup.agi is executable by
the user under which asterisk is running.

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Re: [asterisk-users] A way to check against a list of numbers?

2010-09-10 Thread Roger Burton West
On Fri, Sep 10, 2010 at 03:51:01PM -0500, Hose wrote:
Does anyone have any suggestions as to
how to approach that, or if they have a entirely different way in mind?

AGI script that can look directly at your master list of numbers/routes?

R

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Re: [asterisk-users] What can make G.729a codec hostid change?

2010-09-07 Thread Roger Burton West
On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote:

Note that ifconfig will not necessarily show all of your
interfaces (hard- or soft-) - only the active, configured ones.

ifconfig -a would help here. Kernel upgrades often seem to bring in new
default interfaces.

If this turns out to be the problem, rmmod or a custom kernel
compilation may do the trick. (Of course if you've _lost_ an interface
you were using under etch this may be more of a problem.)

R

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[asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)

2010-09-03 Thread Roger Burton West
I have a pair of Asterisk servers which are happily routeing VoIP calls.
I want to hook one of them to the PSTN. Given that I am in the UK, what
is a reasonably easily-available device to provide an FXO interface from
a Linux box, with a minimum of faffing around with drivers? Just one
line is needed, though in theory two might eventually be useful. My
usual white-box hardware suppliers don't seem to play in this field.

Also: I've heard good things about the PAP2T for getting analogue
handsets to talk to a VoIP server. But all the ones I can see on eBay
are PAP2T-NA models. Will these work with British handsets? (Obviously
with a plug adaptor to put the BT jack into an RJ11 socket, but that's
relatively easy to arrange.)

Roger

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