[asterisk-users] 1.6.2 ConfBridge suggestion

2010-07-16 Thread Steve Johnson
A very nice feature of another conferencing system that I've used is
that the admin/moderator can press a star code to MUTE ALL OTHER USERS
on the conference.

This is great if you have several people on the call and one of the
people puts the call on hold (and so the music/advertisement/your call
is important/etc) message starts, or someone's cellphone handsfree
unit in their car is making a bunch of noise, or someone's endpoint
starts creating acoustic echo.

To minimize disruption on the call, the admin should be able to press
(*5) and this would mute all other users, and then he/she would just
tell the participants that they can unmute their microphone by
pressing *1.

Just a suggestion for whoever is involved with evolving the ConfBridge app.

Regards!

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Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

2010-07-14 Thread Steve Johnson
On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce  wrote:
> Hi Guys,
> Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
> 6730i, but none of them indicate the voic-email. Where should I look for
> trouble to find the root issue for MWI?

(1) Check from the CLI> voicemail show users

to ensure that the proper mailboxes have been set up and there is new
mail in them.  If this is not right, check the voicemail.conf entry
for this mailbox.

(2) Check the phone device configuration (in sip.conf) to ensure that
the phone has a mailbox=xxx entry.

for example:

;entry in sip.conf for extension 115
[115]
context=yourcontext
mailbox=115
...

Restart asterisk if you've made changes and re-test.

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[asterisk-users] Logging codec used in CDR

2010-07-09 Thread Steve Johnson
Happy Friday everyone,

Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?

This is for SIP-based calls, if that matters.

Perhaps there is some variable that can be queried as part of the
dialing script;
Or is it possible to grab the codec name using the "exten =>h," after
the call completes...

Thanks in advance for all suggestions.

S

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[asterisk-users] GoogleTalk to Asterisk - choosing voice menu options

2010-05-27 Thread Steve Johnson
GoogleTalk connects ok to Asterisk 1.6.2.7 but how can you choose
voice menu options (press 1 for Bob, press 2 for Betty, ...) from the
GT client?

(There is no dial pad in the Windows GT client, but what you type in
the message box does show up on the console as an incoming Jabber
message.)

Is there a way? Thanks all!

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[asterisk-users] How to get ConfBridge user count

2010-05-25 Thread Steve Johnson
I want to set up a conference call to be recorded automatically, so
I'd like the recording to start when the second caller joins the
conference (one caller already there).  The recording would continue
until the last user hangs up.

How can you determine how many are already in the conference bridge?

[conferences]
exten => 66,1,Answer
exten => 66,n,Wait(1)
exten => 66,n,Authenticate(123456)
;
exten => 66,n,NoOp(-- ConfBridge 66 user count: ${count} --)
;<-- WHAT VARIABLE TO USE HERE?
;
exten => s,n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
exten => s.,n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
exten => s,n,ExecIf($[${count} =
1]?Monitor(wav,record-${CALLERID(num)}-${DATETIME},bm))
;
exten =>66,n,ConfBridge(66,Ms)
exten => 66,n,Playback(goodbye)
exten => 66,n,Hangup

Thanks for any info.

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[asterisk-users] [Asterisk-Users] Asterisk transfer to a conference using feature code?

2010-05-20 Thread Steve Johnson
Is it possible to use an Asterisk feature code to transfer a call to a
specific extension?

For instance, if you take a call, and the caller wants to go to a
conference, it would be nice to use a feature code for this, rather
than going through a longer transfer sequence.

e.g.:
- You have a meetme conference:
[conferences]
exten => 21,1,NoOp(MeetMe Conference)
exten => 21,n,MeetMe(50,pM)  ;p=prompt for pin, M=music for first caller
exten => 21,n,Hangup

- You then want to define a feature code *5 in features.conf which
will blind transfer the caller to (conferences,21,1)

Any suggestions/examples as to how to set this up?

Thanks

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[asterisk-users] Asterisk Call Recording *1 Status Indication

2010-05-13 Thread Steve Johnson
When you press *1 in Asterisk (1.6.2.7) to start/stop call recording,
the console CLI> shows:

> User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m

Is it possible to play a sound to back to the person who pressed *1 to
indicate to them that recording has actually started or stopped?
Something like "Recording" / "Record Off", or else sounds like people
are used to hearing when they plug/unplug a USB device into a PC.

Also, I'd also like to have the completed recording go to the person's
voicemail box as a message if that's possible when the recording stops
by toggling *1 or the parties hang up, if anyone has suggestions for
doing that or can point to a link.

Thanks much!  S.

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[asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]

2010-04-20 Thread Steve Johnson
On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards
 wrote:
> On Tue, 20 Apr 2010, Tilghman Lesher wrote:
>
>> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
>>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
>>> prompt, and found references on using the command "soft hangup
>>> ", but as you can see below, the "soft hangup" command
>>> does not seem to exist, and there is no mention about it in the
>>> UPGRADE*.txt documents.
>>>
>>> Can anyone shed light on what would replace "soft hangup" in 1.6.2.x ??
>>> (This asterisk server is strictly SIP/IAX2, no DAHDI hardware)
>>
>> "channel request hangup "
>
> How obvious.
>
> Kind of makes me wish I still used 1.2 -- oh wait, I do.
>
> Seriously though, IMNSHO, with every release the CLI gets more obtuse.
>
> I'd like to see a more natural and intuitive interface following a "verb
> noun" model like Oracle, MySQL, or even GDB.
>
>       hangup [sip|iax|dahdi] channel 
>
> seems so "obvious."
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Further to Steve Edward's comment, I think things would be more
obvious if the help system was improved slightly, for instance:

If you were trying to figure out the commands dealing with peers, you
would be able to type:
*CLI> help peer
No "peer" command found.  Possible alternatives:
iax2 show peer Show details on specific IAX peer
   iax2 show peers List defined IAX peers
sip show peers List defined SIP peers
 sip show peer Show details on specific SIP peer
  (and so on, maybe using the "[More]" option to help it be readable)

In this case, if I could use the "help" system to search on all
occurrences of the word "hangup" in the available commands, I would
probably have found it myself instead of bothering the list.

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[asterisk-users] 1.6.2 No "soft hangup"?

2010-04-20 Thread Steve Johnson
Hello asteriskers,

I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
prompt, and found references on using the command "soft hangup
", but as you can see below, the "soft hangup" command
does not seem to exist, and there is no mention about it in the
UPGRADE*.txt documents.

Can anyone shed light on what would replace "soft hangup" in 1.6.2.x
??  (This asterisk server is strictly SIP/IAX2, no DAHDI hardware)

Thanks!


Ref: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg185265.html

Asterisk 1.6.2.7-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
..
*CLI> sip show channels
Peer             User/ANR         Call ID          Format
Hold     Last Message    Expiry
1.2.3.4     104              b45f77d8-f7789d  0x100 (ulaw)     No       Rx: ACK
5.6.7.8   1866***      37e7735e0a294a2  0x4 (ulaw)       No       Rx: ACK

*CLI> core show channels
Channel  Location State   Application(Data)
SIP/isp-00 (None)   Up  AppDial((Outgoing Line))
SIP/104-0012 1866...@outbound: Up  Dial(SIP/isp/1866***
2 active channels
1 active call

*CLI> soft hangup SIP/104-0012
No such command 'soft hangup SIP/104-0012' (type 'core show help
soft hangup' for other possible commands)

*CLI> core soft hangup SIP/104-0012
No such command 'core soft hangup SIP/104-0012' (type 'core show
help core soft' for other possible commands)

*CLI> sip soft hangup SIP/104-0012
No such command 'sip soft hangup SIP/104-0012' (type 'core show
help sip soft hangup' for other possible commands)

*CLI> core stop now
(This stopped the call of course, but also killed asterisk in the process)

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Re: [asterisk-users] Inquiry:Asterisk Dictate?

2009-12-30 Thread Steve Johnson
Google is your friend. You should use it. Search for:
asterisk extensions.conf dictate
or
asterisk extensions.conf dictate example

Some results:
http://www.asteriskguru.com/tutorials/dictate.html
and
   http://www.voip-info.org/wiki/view/Asterisk+cmd+Dictate


On Wed, Dec 30, 2009 at 6:36 AM, hadi motamedi  wrote:
> Dear All
> Can you please give me more hint on how Asterisk Dictate() works?
> Thank you
>
>
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Re: [asterisk-users] Ringing for incoming call

2009-12-18 Thread Steve Johnson
If you try just this, what does the caller hear? It should be ringing
for the first 20 sec, and then maybe the congestion tone afterwards.
exten => s,1,Wait(20)
exten => s,n,Hangup

You shouldn't need/use the Ringing() command at all, as the initial
ring before your system answers would be generated by the provider.

If "wait ... answer" doesn't work for you, you'll have to provide more
output from the CLI and tell us more about your configuration.


On Fri, Dec 18, 2009 at 10:29 PM, Bob Smither  wrote:
>
> On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
>> Try putting the wait before the Answer.
>>
>> ...
>> exten => s,n,Wait(10)
>> exten => s,n,Answer
>> ...
>
> Thanks Steve.  I tried that:
>
>> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
>> > Dear All,
>
> 
>
>> >
>> > ...
>> > exten => s,n,Answer
>> > exten => s,n,Ringing()
>> > exten => s,n,Wait(10)
>> > exten => s,n,BackGround(sound file)
>> > ...
>> >
>> > I have also tried moving the Answer app to right before the BackGround
>> > app.
>
> 
>
> i.e., after the Wait, but still no joy.
>
> Anything else I need to look at?
>
> Thanks,
> --
> Bob Smither 
>
>
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Re: [asterisk-users] Ringing for incoming call

2009-12-18 Thread Steve Johnson
Try putting the wait before the Answer.

...
exten => s,n,Wait(10)
exten => s,n,Answer
...

On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
> Dear All,
>
> I am using Asterisk 1.4 on CentOS 5.  I have an incoming DID provided by
> Vitelity.  When the number is called it goes to my Asterisk box.  The
> protocol is SIP.  This all works just fine if I answer the call and
> begin a playback.
>
> I want to let the number ring for a few seconds before it is answered,
> and would like the caller to hear it ringing.  I have tried:
>
> ...
> exten => s,n,Answer
> exten => s,n,Playtones(ring)
> exten => s,n,Wait(10)
> exten => s,n,StopPlaytones()
> exten => s,n,BackGround(sound file)
> ...
>
> also
>
> ...
> exten => s,n,Answer
> exten => s,n,Ringing()
> exten => s,n,Wait(10)
> exten => s,n,BackGround(sound file)
> ...
>
> I have also tried moving the Answer app to right before the BackGround
> app.
>
> In all cases when I call the number I never hear it ringing.  After the
> 10 second delay, the BackGround app does run.  Connecting to the CLI
> does not give me any useful information - for example the Ringing app is
> shown to run, but the caller does not hear it.
>
> Any suggestions?
>
> Many thanks!
>
> --
> Bob Smither 
>
>
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Re: [asterisk-users] Rewrite calling number of incoming call

2009-12-15 Thread Steve Johnson
How about:

exten => 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten => 977,n,ExecIf($[${CALLERID(num)} = 1234]?Set(CALLERID(num)=317998977))
exten => 977,n,ExecIf($[${CALLERID(num)} = 5678]?Set(CALLERID(num)=317998978))
[..]
exten => 977,n,Dial(SIP/0317998977)


On Mon, Dec 14, 2009 at 12:21 PM, Magnus Benngård
 wrote:
> Hi!
>
> Trying to figure out how to rewrite calling number of an incoming call...
>
> A cell phone (0733025975) dials a X-Lite (977).
> X-Lite "shows" 733025975 at the display, but I want it to be 0317998975.
> I thought i could do something like:
>
> exten => 977/733025975,1,Set(CALLERID(number)=0317998975)
> exten => 977,n,Dial(SIP/0317998977)
>
> [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272 answer_call: Dropping call
> because extensions '977', 's' and 'i' doesn't exists in context
> [inputinterior.se]
>
> Rewriting of outgoing is working... snip
>
> exten => _0X!/0317998975,1,Set(CALLERID(number)=317998975)
> exten => _0X!/0317998977,1,Set(CALLERID(number)=317998977)
> exten => _0X!/0317998978,1,Set(CALLERID(number)=317998978)
> exten => _0X!/0317998985,1,Set(CALLERID(number)=317998985)
> exten => _0X!/0317998987,1,Set(CALLERID(number)=317998987)
> exten => _0X!,n,Dial(H323/0${ext...@avaya)
>
> Can someone guide me on the correct track?
>
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Re: [asterisk-users] G729 with IAX

2009-12-08 Thread Steve Johnson
Of course, as long as your endpoints support it.  Read more about it
and purchase G.729 channel licenses for Asterisk from Digium:

http://www.digium.com/en/products/g729codec.php

Once you have the codec properly installed, enable it for your peer in
your iax.conf file "allow=g729".  Restart asterisk and go to it.

Also, Google is your friend. Search: g729 iax
for lots of information and examples.


On Tue, Dec 8, 2009 at 1:03 AM, DHAVAL INDRODIYA
 wrote:
> dear All,
>
> can I use G729 with IAX trunk or IAX calls
>
> regards
> Dhaval
>
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Re: [asterisk-users] Automon -> Voicemail

2009-12-07 Thread Steve Johnson
Thanks.  One more feature/automon question:

Is there some way to provide audible feedback (play a specific sound
file) to the initiator that a feature has been activated -- in this
case that recording has started (like "bee-yup") and stopped
("bee-dah"), when you enter in the *1?




On Mon, Dec 7, 2009 at 2:58 PM, Doug Lytle  wrote:
> Steve Johnson wrote:
>> Hi all,
>>
>> What's the best method to send automon call recordings (*1) to the
>> voicemail box of the Asterisk user?
>>
>>
>
> I've picked up the following off the list a while ago.  Works pretty
> good.  I do a mysql lookup to see if the user has the ability or not:
>
> __features.conf:__
>
>
> [applicationmap]
>
> recordtovm => *8,self,Macro,recordtovm
>
>
> __Dial plan entry:__
>
> ; 
> ; Call recording, initiated by *8
> ; after hangup, send recording to
> ; callers voice mail box
> ; 
>
> [macro-recordtovm]
>
> exten => s,1,MYSQL(Connect connid localhost username 'supersecret'
> call_recording)
> exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
> exten => s,n,MYSQL(Query resultid ${connid} SELECT allowed FROM
> Indianapolis WHERE extension = ${CALLERID(number)})
> exten => s,n,MYSQL(Fetch fetchid ${resultid} results)
> exten => s,n,MYSQL(Disconnect ${connid})
> exten => s,n,MYSQL(Clear ${resultid})
> exten => s,n,Set(RECORDING.OK=${results})
> exten => s,n,GotoIf($["${results}" = "Y"]?9:15)
> exten =>
> s,n,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
> exten => s,n,Set(ORIG_DATE=${STRFTIME(${EPOCH},,%c)})
> exten => s,n,Set(ORIG_TIME=${STRFTIME(${EPOCH},,%s)})
> exten => s,n,Set(ORIG_CID=${ARG2})
> exten => s,n,Playback(local/stutter)
> exten =>
> s,n,MixMonitor(${MONITOR_FILENAME}.wav,b,/usr/local/bin/recordtovm.pl
> ${CALLERID(num)} ${MONITOR_FILENAME}.wav "${ORIG_DATE}" "${ORIG_TIME}"
> "${ORIG_CID}")
> exten => s,n,NoOP(User allowed to record calls? ${results})
>
>
>
> __Perl script__
>
> cd /usr/local/bin
>
> cat recordtovm.pl
>
> #!/usr/bin/perl -w
> #
> use strict;
>
> my $monitordir="/var/spool/asterisk/monitor/";
> my $vmdir="/var/spool/asterisk/voicemail/sip/";
> my $vmfolder="INBOX";
> my $vmbox=$ARGV[0];
> my $vmpath=$vmdir."$vmbox/"."$vmfolder";
> my $monitorfilename=$ARGV[1];
> my $orig_date=$ARGV[2];
> my $orig_time=$ARGV[3];
> my $orig_cid=$ARGV[4];
>
> opendir (DIR, $vmpath);
> my @files = grep(/\.txt$/,readdir(DIR));
> closedir(DIR);
> my @sortedfiles = sort {$b cmp $a} @files;
> my $vmid;
> if ($sortedfiles[0] =~ /^(msg)(\d\d\d\d)(.txt)/)
> {
>     $vmid=$2;
>     $vmid++;
> }
> else
> {
>     $vmid="";
> };
>
> open VMFILE,"> $vmpath/msg$vmid.txt";
> print VMFILE ";\n";
> print VMFILE "; Message Information file\n";
> print VMFILE ";\n";
> print VMFILE "[message]\n";
> print VMFILE "origmailbox=$vmbox\n";
> print VMFILE "context=\n";
> print VMFILE "macrocontext=\n";
> print VMFILE "exten=s\n";
> print VMFILE "priority=\n";
> print VMFILE "callerchan=\n";
> print VMFILE "callerid=$orig_cid\n";
> print VMFILE "origdate=$orig_date\n";
> print VMFILE "origtime=$orig_time\n";
> print VMFILE "category=\n";
> print VMFILE "duration=\n";
> close VMFILE;
>
> if ($ARGV[1])
> {
>     system("mv $monitordir"."$monitorfilename $vmpath/msg$vmid.wav");
> };
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
>
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[asterisk-users] Automon -> Voicemail

2009-12-07 Thread Steve Johnson
Hi all,

What's the best method to send automon call recordings (*1) to the
voicemail box of the Asterisk user?

Do you have to trap hangups, etc, or is there some global variable
that can be set?

Thanks!

S.

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Re: [asterisk-users] local channels

2009-11-09 Thread Steve Johnson
>  My Dial() command is Dial($LOCAL_DIAL)

Perhaps you should be using:

 Dial(${LOCAL_DIAL})

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[asterisk-users] Determining extension's sip.conf default mailbox

2009-10-31 Thread Steve Johnson
Hello list,

How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with "sip show peer ")?  Is there a
function to extract it?

Why?  Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
mailbox as coded in sip.conf.

sip.conf
--
[100]
mailbox=100

[102]
mailbox=102

[103]
mailbox=100

I want a function which I can use in the dialplan (1.6) that works like:
DefaultMailbox(100) -> 100
DefaultMailbox(102) -> 102
DefaultMailbox(103) -> 100

for example:
exten s,n,VoicemailMain(DefaultMailbox(${CALLERID(num)}))

Suggestions?
Thanks!

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

2009-05-26 Thread Steve Johnson
For long distances, a wireless point-to-point might be more economical
than trenching.

e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender
http://www.oksolar.com/communications/phone_line_ext.htm

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Re: [asterisk-users] Where are 2 letter language values defined?

2009-05-06 Thread Steve Johnson
Also check out:
http://www.w3.org/International/questions/qa-lang-2or3.en.php


On Wed, May 6, 2009 at 1:22 PM, Steve Johnson  wrote:
> They are 2-letter ISO country codes.
>
> http://www.iso.org/iso/english_country_names_and_code_elements
>
> On Wed, May 6, 2009 at 1:05 PM, Steve Edwards  
> wrote:
>> I've googled for way too long, where are the 2 letter language values
>> defined?
>>
>> I know:
>>
>> en = English
>> es = Spanish
>> fr = French
>>
>> but what about Croatian, Russian, Serbian, Vulcan, etc?
>>
>> Is there a list documented for Asterisk or is it "just use the 2 letter
>> country code Internet TLD?"
>>
>> Thanks in advance,
>> 
>> Steve Edwards      sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
>> Newline                                             Fax: +1-760-731-3000
>>
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Re: [asterisk-users] Where are 2 letter language values defined?

2009-05-06 Thread Steve Johnson
They are 2-letter ISO country codes.

http://www.iso.org/iso/english_country_names_and_code_elements

On Wed, May 6, 2009 at 1:05 PM, Steve Edwards  wrote:
> I've googled for way too long, where are the 2 letter language values
> defined?
>
> I know:
>
> en = English
> es = Spanish
> fr = French
>
> but what about Croatian, Russian, Serbian, Vulcan, etc?
>
> Is there a list documented for Asterisk or is it "just use the 2 letter
> country code Internet TLD?"
>
> Thanks in advance,
> 
> Steve Edwards      sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                             Fax: +1-760-731-3000
>
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[asterisk-users] one-button call parking/pickup on Asterisk with Polycom phones?

2009-04-10 Thread Steve Johnson
Anyone want to talk briefly about one-button call parking/pickup on
Asterisk with Polycom phones? Does anyone use it or know to do it?

On many phone systems there are 2 or 3 park buttons, and you can park
a call onto an unlit park button, and then the light flashes.  You can
go to any other phone, and press the park button with the flashing
light to pick up the call.

Super easy from the user's point of view.

How to do with Asterisk? Hints?

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[asterisk-users] Patch to dahdi Chans.pm

2009-02-02 Thread Steve Johnson
Software:
dahdi-linux-complete-2.1.0.3+2.1.0.2.tar.gz
asterisk-1.6.1-rc1.tar.gz

Hardware:
4-port fxs card

Example:
# /etc/init.d/dahdi status

### Span  1: WRTDM/0 "wrtdm Board 1" (MASTER)
  1 FXSFXSKS   (In use)
  2 FXSFXSKS   (In use)
  3 FXSFXSKS   (In use)
  4 FXSFXSKS   (In use)
Use of uninitialized value in string eq at
/usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221.
  5 unknown
Use of uninitialized value in string eq at
/usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221.
  6 unknown
Use of uninitialized value in string eq at
/usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221.
  7 unknown
[..]
Use of uninitialized value in string eq at
/usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221.
 23 unknown
Use of uninitialized value in string eq at
/usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221.
 24 unknown

Problem:
$self->type is not being checked to see if it is defined.  Add a line
just above line 221 to fix this, such as:

return undef unless defined $self->type;

After the fix, the above command executes without error.

# /etc/init.d/dahdi status
### Span  1: WRTDM/0 "wrtdm Board 1" (MASTER)
  1 FXSFXSKS   (In use)
  2 FXSFXSKS   (In use)
  3 FXSFXSKS   (In use)
  4 FXSFXSKS   (In use)
  5 unknown
  6 unknown
  7 unknown
  8 unknown
  9 unknown
 10 unknown
 11 unknown
 12 unknown
 13 unknown
 14 unknown
 15 unknown
 16 unknown
 17 unknown
 18 unknown
 19 unknown
 20 unknown
 21 unknown
 22 unknown
 23 unknown
 24 unknown

FYI...

S.

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Re: [asterisk-users] Voicemail message is dialtone

2009-01-16 Thread Steve Johnson
Here's also an example snip from the debug log:

[07:42:20] -- Executing [...@mainmenu:15] Dial("Zap/1-1",
"SIP/105|18|tKk") in new stack
[07:42:20] -- SIP/105-08571180 is ringing
[07:42:39] -- Nobody picked up in 18000 ms
[07:42:39] -- Executing [...@mainmenu:16] Answer("Zap/1-1", "") in new stack
[07:42:39] -- Executing [...@mainmenu:17] Wait("Zap/1-1", "1") in new stack
[07:42:40] -- Executing [...@mainmenu:18] Playback("Zap/1-1",
"silence/1") in new stack
[07:42:40] --  Playing 'silence/1' (language 'en')
[07:42:41] -- Executing [...@mainmenu:19] BackGround("Zap/1-1",
"please-leave-a-message") in new stack
[07:42:41] --  Playing 'please-leave-a-message' (language 'en')
[07:42:47] -- Executing [...@mainmenu:20] VoiceMail("Zap/1-1", "105|s")
in new stack
[07:42:48] --  Playing 'beep' (language 'en')
[07:42:48] -- Recording the message
[07:42:48] -- x=0, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav49,
0x8570eb8
[07:42:48] -- x=1, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: gsm,
0x8227430
[07:42:48] -- x=2, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav,
0x8259b10
[07:46:42] -- Recording automatically stopped after a silence of 10 seconds
[07:46:42] --  Playing 'auth-thankyou' (language 'en')
[07:46:43] == Auto fallthrough, channel 'Zap/1-1' status is 'NOANSWER'
[07:46:43] -- Hungup 'Zap/1-1'


On Fri, Jan 16, 2009 at 12:07 PM, Steve Johnson  wrote:
> Hello all,
>
> I have one Asterisk 1.4.21 system connected to a North American POTS
> line.  Normally hangup detection works fine, and Asterisk hangs up
> properly if you are talking to a caller and they hang up; but
> occasionally a call comes in (typically from a US telemarketer) where
> the caller hangs up just as voicemail recording is starting, and you
> get a voicemail recording of dialtone (then congestion and off-hook
> warning tones) for almost 4 minutes before asterisk gives up the line.
>
> zapata.conf [channels] options are:
>
> language=en
> context=default
> rxwink=300
> usecallerid=yes
> hidecallerid=no
> cidsignalling=bell
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=no
> transfer=yes
> canpark=yes
> cancallforward=no
> callreturn=no
> musiconhold=default
> echocancel=yes
> echocancelwhenbridged=yes
> immediate=no
> faxdetect=no
> relaxdtmf=yes
> hanguponpolarityswitch=yes
> progzone=us
> signalling=fxs_ks
> channel => 1
>
>
> Any suggestions for voicemail detecting/rejecting messages when there
> is only dialtone on the other end?
>
> Thanks!
>
> S.
>

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[asterisk-users] Voicemail message is dialtone

2009-01-16 Thread Steve Johnson
Hello all,

I have one Asterisk 1.4.21 system connected to a North American POTS
line.  Normally hangup detection works fine, and Asterisk hangs up
properly if you are talking to a caller and they hang up; but
occasionally a call comes in (typically from a US telemarketer) where
the caller hangs up just as voicemail recording is starting, and you
get a voicemail recording of dialtone (then congestion and off-hook
warning tones) for almost 4 minutes before asterisk gives up the line.

zapata.conf [channels] options are:

language=en
context=default
rxwink=300
usecallerid=yes
hidecallerid=no
cidsignalling=bell
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=yes
canpark=yes
cancallforward=no
callreturn=no
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
immediate=no
faxdetect=no
relaxdtmf=yes
hanguponpolarityswitch=yes
progzone=us
signalling=fxs_ks
channel => 1


Any suggestions for voicemail detecting/rejecting messages when there
is only dialtone on the other end?

Thanks!

S.

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Re: [asterisk-users] Variables for dial plan

2008-12-15 Thread Steve Johnson
One of these methods will work:

exten => s,n,ExecIf($["${dialplan}" = "NZ"]|Set|NAT="0")
exten => s,n,ExecIf($["${dialplan}" = "NZ"]|Set|INT="00")

-or-

exten => s,n,GotoIf($["${dialplan}" != "NZ"]?not-nz)
exten => s,n,Set(NAT="0")
exten => s,n,Set(INT="00")
exten => s,n(not-nz),more_dialplan_stuff


On Mon, Dec 15, 2008 at 3:26 AM, Michael  wrote:
> On Mon, 15 Dec 2008 21:31:56 you wrote:
>> Use setvar=variablename=value
>>
>> Eg: under [client1]
>> setvar=dialplan=NZ
>>
>> Then just reference ${dialplan} in your extensions.conf
>>
>> Cheers
>> Andy
>
> Thanks, now how do I achieve the following logic?
>
> if ($dialplan == NZ) {
> $NAT = 0;
> $INT = 00;
> };
>
> Michael
>
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[asterisk-users] Park buttons on Polycom IP501/601

2008-12-10 Thread Steve Johnson
Is anyone using fixed Park buttons (some of the ones on the left side
of the screen) on a Polycom phone?  Here's what I mean:

- Call is received and parked, by the user pressing an unlit park
button (e.g. 701) and the call is parked there.
- The call can be picked up at any other extension by pressing the
flashing park 701 button.
- Once the call has been picked up, the 701 park slot is idle and the
light goes off.

For a small site, only a couple of Park buttons would be needed.

Can you give an example of how to do this?

Thanks,
S.

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[asterisk-users] Alternate names in Directory (dial-by-name)

2008-05-16 Thread Steve Johnson
Hi everyone,

What creative methods are used to support dial-by-name functionality
for people who go by more than one name?

e.g.:   Rebecca/Becky, Margaret/Peggy, William/Bill, Liz/Elizabeth, etc.

We'd like to use the "f" first name option of the Directory function,
as the particular phone system has multiple members of the same
families (same last names).

In the first example, the caller should be able to key either R-E-B or
B-E-C and get to the same person/mailbox.

Any suggestions?

Thanks,
S.

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[asterisk-users] VOICEMAIL OPTIONS help needed

2008-05-07 Thread Steve Johnson
Hi everyone,

We have a particular user on our Asterisk 1.4.x system who always
listens to his voicemail messages via email.

- Is there some way to send the voicemail ONLY to email and not retain
them on the phone?

- Alternatively, can the voicemail system only keep, say, just the
last 10 messages (as backup in case of email delivery failure or a
message getting deleted in email accidentally before it is heard),
purging out the oldest when a new one is received?
(If we set the option maxmsg=10 on his mailbox in voicemail.conf, I
think it will stop accepting voicemails after 10 messages, not turf
the oldest one and accept a new one in its place).

Everyone else uses the normal voicemail options on their phones, so
the solution should be just for this single user.


Thanks for any suggestions.

S.

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Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones

2008-03-28 Thread Steve Johnson
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales <[EMAIL PROTECTED]> wrote:
>
>  Can't you just use the same bootrom for all your polycom phones?
>
>  PaulH
>
>
>
>
>  On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
>  > I have a question about DHCP and boot server supporting more than 1
>  > model of Polycom phones.
>  >
>  > According to Polycom "standards", Polycom phone boots up to get a DHCP
>  > address and at the same time getting a boot server string (with username
>  > and password) to logon to boot server to download SIP, bootROM and etc.
>  >
>  > That is okay if there is only one type of phone (that requires a
>  > specific SIP and bootROM release).
>  >
>  > What about if the environment has to support two or more models of
>  > Polycom phones?
>  >
>  > On the boot server side, I can define another home directory like
>  > /home/polycom1 and /home/polycom2 to store different SIP and bootROM
>  > releases.  However, the issue is how different polycom phone model can
>  > get a different user account and password to log on to different home
>  > directories.
>  >
>  > I understand the issue here is DHCP and not the boot server but I am a
>  > bit of a newbie here.
>  >
>  > Can anyone help please?
>  >

As someone earlier pointed out, different models of polycom phones can be
pointed to the same set of configuration files.  With the standard ISC dhcpd
server, the phones can be told where to look by using a directive like:

option tftp-server-name "ftp://polycom:[EMAIL PROTECTED]/";

This would require a user account on the ftp server like:

polycom:x:501:501:Polycom Phone
Provisioning:/etc/asterisk/polycom/ftp/:/bin/bash

and the configuration files would be placed in the /etc/asterisk/polycom/ftp/
directory.
So if you wanted to have separate configurations for certain phones (for
upgrade testing, etc., it is easily possible.

SJ

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Re: [asterisk-users] is encrypted iax safe and secure?

2008-02-15 Thread Steve Johnson
Of course *it would be nice if* the IAX2 authentication parameters
were also encrypted, so that there was no danger of a 3rd party
hijacking your connection and generating a bunch of extra charges.

S.

On Fri, Feb 15, 2008 at 11:31 AM, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Tim Panton wrote:
>
>  > The NEW frame doesn't _have_ to contain a dialed number, the digits
>  > can be sent later
>  > (I forget the frametype), but later means within the encrypted
>  > session :-)
>
>  It's the DIAL command that you are thinking of. I'm considering
>  implementing this, but it has one major caveat: to really do the job
>  right, we wouldn't want any caller information (CLID or CNAM) to be in
>  the NEW message either, it would have to be added as IEs to the DIAL
>  command. Unfortunately no existing implementations are going to be
>  prepared to receive that information as part of DIAL, so they would
>  process this sort of call with an empty CLID and CNAM. We can of course
>  enhance chan_iax2 to understand this method of doing things, but it
>  won't be backward compatible with previous versions of Asterisk or any
>  other IAX2 clients.
>
>  --
>  Kevin P. Fleming
>  Director of Software Technologies
>  Digium, Inc. - "The Genuine Asterisk Experience" (TM)
>
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Re: [asterisk-users] Help: dtmf mode

2008-01-24 Thread Steve Johnson
Please post your sip.conf entry for your phone and also describe your
calling path.  Are you having a problem with internal calls (e.g.: to
voicemailmain) on the same switch, or are you referring to calls to
PSTN destinations via pots/pri/sip/?  Also, which versions of
Asterisk, Zaptel, linux, etc. are you using?

S.

On Jan 24, 2008 12:43 PM, Jarga Jallow <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Hi,
>
> I am having trouble making a selection when I call a number and need to make
> a selection to go to an extension with my polycom phones 301. Anybody have
> an idea how to fix this problem?
>
> Thanks in advance.
>
>
>
>
> Jarga Jallow
>
> Technical Support Engineer
>
> 2985 S. Hwy. 360
>
> Grand Praire, Texas 75052
>
> Direct: 972-206-1212 ext# 29
>
> Mobile: 214-669-9046
>
> Fax:972-999-4113
>
> Toll Free: 1-877-801-5511 ext 34
>
> Toll Free: 1-877-926-2288
>
>
>
> www.2mcctv.com
>
>
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Re: [asterisk-users] Polycom-SIP response 500

2008-01-22 Thread Steve Johnson
I have just retested and agree that this error eventually does clear
itself.  However, in this test it took about 35 minutes and each
Polycom phone produced between 1000 and 1300 error message lines at 1
to 0 second intervals (which I captured to the debug log).  Once one
phone starts flagging an error, all Polycom phones that are
buddy-watching join right in.

I triggered the problem by simply restarting asterisk:
/usr/sbin/asterisk -rx "restart when convenient".  Sometimes (but not
all the time) it will also start if you reload.

All suggestions appreciated.

S.

On Jan 22, 2008 9:23 AM, Steve Johnson <[EMAIL PROTECTED]> wrote:
> I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
> this.  It seems to occur less often with "extensions reload" rather
> than just "reload", but it would be nice to fix this.
>
> Tx.
>
>
>
> On Jan 22, 2008 8:30 AM, Steve Davies <[EMAIL PROTECTED]> wrote:
> > On 1/22/08, Steve Johnson <[EMAIL PROTECTED]> wrote:
> > > Hi list,
> > >
> > > There are many Polycom experts on this list -- hopefully someone has a 
> > > solution.
> > >
> > > With *several* versions of Asterisk 1.4.x, doing a reload  of Asterisk
> > > causes the Polycom 601 phones to start dumping these messages to the
> > > CLI.
> > >
> > > -- Incoming call: Got SIP response 500 "Internal Server Error"
> > > back from 192.168.2.x
> >
> > [snip]
> >
> > I have not seen this problem here since upgrading to 2.1.2 firmware.
> > Or perhaps it was 2.2.0, one or the other. The phones now seem to
> > recover on thier own when Asterisk returns.
> >
> > Cheers,
> > Steve
> >
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Re: [asterisk-users] Polycom-SIP response 500

2008-01-22 Thread Steve Johnson
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
this.  It seems to occur less often with "extensions reload" rather
than just "reload", but it would be nice to fix this.

Tx.


On Jan 22, 2008 8:30 AM, Steve Davies <[EMAIL PROTECTED]> wrote:
> On 1/22/08, Steve Johnson <[EMAIL PROTECTED]> wrote:
> > Hi list,
> >
> > There are many Polycom experts on this list -- hopefully someone has a 
> > solution.
> >
> > With *several* versions of Asterisk 1.4.x, doing a reload  of Asterisk
> > causes the Polycom 601 phones to start dumping these messages to the
> > CLI.
> >
> > -- Incoming call: Got SIP response 500 "Internal Server Error"
> > back from 192.168.2.x
>
> [snip]
>
> I have not seen this problem here since upgrading to 2.1.2 firmware.
> Or perhaps it was 2.2.0, one or the other. The phones now seem to
> recover on thier own when Asterisk returns.
>
> Cheers,
> Steve
>
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[asterisk-users] Polycom-SIP response 500

2008-01-21 Thread Steve Johnson
Hi list,

There are many Polycom experts on this list -- hopefully someone has a solution.

With *several* versions of Asterisk 1.4.x, doing a reload  of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.

-- Incoming call: Got SIP response 500 "Internal Server Error"
back from 192.168.2.x

They continue on until we force the devices to reboot from the CLI
with a "sip notify polycom-check-cfg 140 141 142 ..." command.

Of course this is "unpleasant" especially during the day.

I have scoured the archives, google and the wiki and have found that
although many experience this problem, no one has posted a solution.

Help please! Thanks much!

Steve

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Re: [asterisk-users] SIPAddHeader in .call file

2008-01-19 Thread Steve Johnson
Sorry to answer my own post, but I have found a solution which perhaps
others can use too...

In the .call file, instead of specifying a channel line as:

  chan: SIP/140  (for example)

use instead:

  chan: Local/[EMAIL PROTECTED]

and put in extensions.conf

[polycom-paging]
exten => _1XX,1,NoOp(Paging Ext ${EXTEN})
exten => _1XX,n,SIPAddHeader(Alert-Info: Ring Answer)
exten => _1XX,n,Dial(SIP/${EXTEN},20,L(6))
exten => _1XX,n,Hangup


Steve Johnson wrote:
> Hi everyone,
>
> How can I add the equivalent of:
>
>exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)
>
> in a .call file?  This is to support paging to Polycom phones...
>
> Thanks for all info!
>
> Steve
>

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[asterisk-users] SIPAddHeader in .call file

2008-01-19 Thread Steve Johnson
Hi everyone,

How can I add the equivalent of:

   exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)

in a .call file?  This is to support paging to Polycom phones...

Thanks for all info!

Steve

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Re: [asterisk-users] Asterisk 1.4 Call Recording

2008-01-14 Thread Steve Johnson
Here's what I would suggest. You should insert some NoOp() statements
and watch the CLI as you dial your 555 extension so that you can see
whether it's working or not.

Your example (which you mentioned you want to run under Asterisk 1.4):
> [test]
> exten => 
> 555,1,SetVar(CALLFILENAME=outgoing/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${EXTEN})
> exten => 555,2,Monitor(wav,${CALLFILENAME},m)
> exten => 555,3,Dial(IAX2/ics.iax-trunk/${EXTEN})
> exten => 555,4,Hangup()

In Asterisk 1.4 (read the UPGRADE.txt file in the source directory):
- SetVar() has been replaced by Set()
- ${TIMESTAMP} no longer exists.
- (CALLERID usage has changed in 1.4 also).

so you first have to fix that stuff, with something like:


[test]
exten => 555,1,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
exten => 555,n,NoOp(DATETIME: ${DATETIME})
;
; Tweak the callfilename until you're happy with it...
; Note that the default recording directory is /var/spool/asterisk/monitor
exten => 555,n,Set(CALLFILENAME=${CALLERID(num)}-${DATETIME}-${EXTEN})
exten => 555,n,NoOp(CALLFILENAME: ${CALLFILENAME})
;
exten => s,n,Monitor(wav,${CALLFILENAME},b)
;
; Remove this next line after the determining that you have the filename right
; by checking the console progress...
exten => 555,n,Hangup()
;
exten => 555,n,Dial(IAX2/ics.iax-trunk/${EXTEN})
exten => 555,n,Hangup()


My script assumes that the monitor files are in the default directory,
so adjust it if necessary after you get the above working.  When you
run it, the .mp3 stereo files should be produced.

In a production environment, I'd imagine that you'd want to run the
combine.sh script periodically as a scheduled cron job.

Don't forget to follow the legal standards for call recording in your
jurisdiction (and be nice).

Have fun,
S.


On Jan 14, 2008 12:53 PM, Mike Hammett <[EMAIL PROTECTED]> wrote:
> Does what I have in the dialplan look right or am I way off base with being
> able to use that script?
>
>
> -
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>
>
>
> - Original Message -
> From: "Steve Johnson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
> Non-Commercial Discussion" 
> Sent: Monday, January 14, 2008 10:51 AM
> Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
>
>
> > You might take a few ideas from this combine.sh script which works for
> > me.  It uses the combine_wave program from
> > http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame
> > program to convert to mp3.
> >
> > It converts the entire directory /var/spool/asterisk/monitor/*-in.wav
> > files to mp3 where the mp3 file doesn't already exist.
> >
> > S.
> >
> >
> > File: combine.sh
> > ---
> > #!/bin/sh
> >
> > cd /var/spool/asterisk/monitor
> >
> > for f in *-in.wav
> > do
> >in=$f
> >out=`echo $f | sed -e 's/-in.wav/-out.wav/'`
> >tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'`
> >mp3=`echo $f | sed -e 's/-in.wav/.mp3/'`
> >
> >if [ -e "$mp3" ]
> >then
> >continue
> >fi
> >
> ># combine the two tracks into one stereo file
> >/usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2>/dev/null
> >
> >/usr/bin/lame --silent -h -b 96 $tmpwav $mp3
> >
> ># Remove temporary .wav files
> >test -w $tmpwav && rm $tmpwav
> >
> ># Remove input files if successful
> >test -s $mp3 && rm $in $out
> > done
> >
> > exit 0
> >
>
>
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Re: [asterisk-users] Voicemail check

2008-01-14 Thread Steve Johnson
The user will receive email notification if you have configured the
user's email address in /etc/asterisk/voicemail.conf .

See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf

Also check the externnotify option which lets you run an external
script when new voicemail is received.

S.

On Jan 14, 2008 11:52 AM, Gilberto Nunes <[EMAIL PROTECTED]> wrote:
> Hi all
>
> Someone knows how can I do to send any notify to user, when he received a new
> message in your mailbox on voicemail?
>
> Thanks for any help
>
> --
> Gilberto Nunes
>
> Itajaí - SC
>
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Re: [asterisk-users] Asterisk 1.4 Call Recording

2008-01-14 Thread Steve Johnson
You might take a few ideas from this combine.sh script which works for
me.  It uses the combine_wave program from
http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame
program to convert to mp3.

It converts the entire directory /var/spool/asterisk/monitor/*-in.wav
files to mp3 where the mp3 file doesn't already exist.

S.


File: combine.sh
---
#!/bin/sh

cd /var/spool/asterisk/monitor

for f in *-in.wav
do
in=$f
out=`echo $f | sed -e 's/-in.wav/-out.wav/'`
tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'`
mp3=`echo $f | sed -e 's/-in.wav/.mp3/'`

if [ -e "$mp3" ]
then
continue
fi

# combine the two tracks into one stereo file
/usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2>/dev/null

/usr/bin/lame --silent -h -b 96 $tmpwav $mp3

# Remove temporary .wav files
test -w $tmpwav && rm $tmpwav

# Remove input files if successful
test -s $mp3 && rm $in $out
done

exit 0

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Re: [asterisk-users] Polycom 330 beep on new VM

2007-12-21 Thread Steve Johnson
This is pretty easy to suppress using the configuration files.  Check:

http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio


On Dec 21, 2007 11:55 AM, Ugo Bellavance <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have a Polycom 330 that emits a beep every 30s or so when there is a
> message waiting.  Is there a way to disable that?  It is pretty annoying.
>
> Regards,
>
> Ugo
>
>
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[asterisk-users] DNS broken for www.voip-info.org ??

2007-12-15 Thread Steve Johnson
The DNS for www.voip-info.org seems to be non-responsive.  Is there a
mirror of this invaluable resource site?

Tx,
Steve

 dig www.voip-info.org
;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server

; <<>> DiG 9.4.1-P1 <<>> www.voip-info.org
;; global options:  printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 61402
;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0

;; QUESTION SECTION:
;www.voip-info.org. IN  A

;; Query time: 4724 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sat Dec 15 11:54:57 2007
;; MSG SIZE  rcvd: 35

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[asterisk-users] [Asterisk-users] Show calls in progress

2007-12-07 Thread Steve Johnson
Is there an Asterisk CLI> command to show a list of calls in progress
(for all channels: Zap/SIP/IAX2 etc).

"Restart when convenient" waits until the system is idle, but is there
an obvious way of seeing what's going on at the moment?

Thanks,
Steve

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[asterisk-users] Re: Asterisk server and DSCP QOS

2007-12-05 Thread Steve Johnson
Thanks, Darryl,

To clarify:

in /etc/asterisk/sip.conf you have the lines:

tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef   ; Sets TOS for RTP audio packets.

and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you
have something like (this is the one I'm uncertain about):

   
  
 
 
 
  
  
 
 
  
   

Thanks again!
Steve


Darryl Duncan wrote:

We're using 184 here (aka TOS 5/EF).

Not set by iptables though, instead it is set in sip.conf
(tos_sip/tos_audio) and on our Polycom/Cisco phones.

-Original Message-
Subject: [asterisk-users] Asterisk server and DSCP QOS

Can anyone comment on the DSCP quality of service settings on your
Asterisk server?

The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.

What iptables settings are you using on the Asterisk server for DSCP?
What are your Polycom DSCP settings?  We're using the Linksys SGE2000P
POE switch which supports QOS via DSCP.

Thanks a lot for any info.
Steve

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[asterisk-users] Asterisk server and DSCP QOS

2007-12-05 Thread Steve Johnson
Can anyone comment on the DSCP quality of service settings on your
Asterisk server?

The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.

What iptables settings are you using on the Asterisk server for DSCP?
What are your Polycom DSCP settings?  We're using the Linksys SGE2000P
POE switch which supports QOS via DSCP.

Thanks a lot for any info.
Steve

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[asterisk-users] Requiring a login to a phone

2007-12-01 Thread Steve Johnson
Hi List,

We have a remote asterisk SIP phone at the cottage.

I'd like it to have minimal privileges when it first registers with
Asterisk. Ideally it should be in a restricted context.  Dialing any
number would intercept the call and tell the person to log on.  This
way, if the phone was stolen or someone got into the cottage, we
wouldn't have a bunch of surprise charges on our phone bill... :-)

Once the phone has been authenticated, it should go into a context
with normal privileges.  After a couple of days of non-use, it should
auto-logout to the restricted context.

How can I change the sip context of a phone on the fly, based on
authentication login?

Any ideas? Thanks,
Steve

sip.conf:
---
; phone at the cottage
[155]
context=restricted-155
...


extensions.conf


[restricted-155]
exten _X.,1,NoOp(All Calls filter through this if not logged in on 155]
exten _X.,n,Answer
exten _X.,n,Wait(1)
exten _X.,n,Playback(You must log in to use this phone)
exten _X.,n,Authenticate(65535)
// if the person authenticates sucessfully, change the context of ext 155
// from restricted-155 to sip-phones.(HOW???)

[sip-phones]
; normal sip phone outgoing context
...

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