Re: [asterisk-users] Any way of limiting incoming caller connection time without making 2 active calls for each incoming call?

[D Tucny]

Jonathan,

Have you tried:
same => n,Set(TIMEOUT(absolute)=3540)

You could override later if required.

Thanks,

Dan

On 17 Apr 2017 04:49, "Jonathan H"  wrote:

The following setup prevents callers from going over 59 minutes:

--

[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/
time_limit_reached)
same => n,Dial(Local/s@root/n,3,L(354:6))
same => n,Hangup()

[root]

exten => s,1,Verbose(Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,etc etc

--

Thing is, each call shows as 2 calls in the console. Not a big
problem, but also now I want to offer a way of overriding that time
limit.

Is there a different or better way of doing this while still hanging
up after 59 minutes, even when the original incoming call has been
transferred elsewhere?

Thanks.

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Re: [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?

[D Tucny]

On 24 July 2012 15:18, Hans Witvliet aster...@a-domani.nl wrote:

 On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote:
  Any tips on solving the following performance conundrum:
 
  Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
 
  tcpdump running to capture UDP 5060/SIP signaling to .pcap files
 
  All calls are ultimately B2BUA client - asterisk - PSTN
 
  Media stays on Asterisk at all times
 
  AGI script has exit handler that connects and updates an external
  database upon BYE from either side.
 
  I know that if exit handler hangs around too long, Bad Things (tm) will
 happen.
 
  Oddly, under load (60-100 B2BUA calls), the G7s start complaining:
 
  Autodestruct on dialog 'CALLID' with owner in place (Method: BYE)
 
  I/O wait is actually higher on the G5s, the G7s have fancy disk cache
  cards and never get above 1% i/o wait
 
  turn off the tcpdump process on the G7s and Autodestruct warnings go
  away.  The G7s should have
  much more capacity than the G5s but we never, ever get Autodestruct
  ... Method: BYE on the G5s.
 
  OS is identical CentOS in both cases.   Every other environmental
  config is the same (network, subnet, DNS etc).
 
  Architecture/bus/network card difference?  tcpdump starving some other
  resource to cause stuff to slow down?

 Hi Tom,

 Regarding G5's and G7's
 We have them both (all though not for asterisk) at work. and found a
 little snag.
 The purchase department interferred with our initial ordering of the
 hardware. As a concequence, they did not order the battery for the raid
 controller. (seems you have to order that explicitly)
 After it arrived, we were able to install linux on it and use it, but
 disk-io was way slower that the original G5.

 You might want to check/compare disk-io  throughput on your G5 vs G7.
 Just a thought


For reference... In my opinion HP servers should never be bought without
the battery or alternative, they shouldn't even be offered for sale without
it...

Without a battery or non-volatile, flash based, cache the controllers in HP
servers disable write caching allocating 100% of the built in cache to be a
read cache. The cache built into the actual disks is also disabled in all
cases, even with a battery. Both of these things are done to protect data.
In the event that a write was in any of the caches during a loss of power,
that data would be lost, a problem made worse by the fact that when data is
written to cache the OS is told it's been safely written. HP refuse to let
you put your data at risk, refusing to activate write caching without a
charged battery attached or NV cache.

I personally would like to have the option to override things like this at
my own risk, but, HP don't give you that option.

With all write caching disabled writes become very slow. The disks can't do
their own reordering of writes to maximise performance. The controller
can't do it's own write optimisations. If you use RAID 5/6, you'll get even
more dire performance thanks to a combination of an underpowered controller
in most cases, plus the fact that writes will often need to read data to
complete the parity write, something that the controller can do in a
smarter, not necessarily serial, way if it's got a cache of things that
need to be written.

So, in summary, HP servers with no battery or nv cache typically have
horrendous disk performance that can't be fixed without adding a battery or
nv cache.

In this case, the fact that the OP mentioned that the io-wait is a lot
lower on the G7s would suggest that they potentially do have a battery or
nv cache which has triggered the controller to enable the write cache
leading to a much improved write performance over what they would have
without, and potentially, the G5s don't have this.

I'd be interested to see if other heavy writes caused the same problem and
also would be interested in more detailed server specs...

d
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Re: [asterisk-users] Realtime Issue

[D Tucny]

How about...

exten = _X.,1,NoOp()

exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})

exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})

exten = _X.,4,Set(usernamepair=${CUT(NULL,\,,1)})

exten = _X.,5,Set(username=${CUT(usernamepair,=,2)})

exten = _X.,6,NoOp(DEVICE is ${DEVICE})

exten = _X.,7,NoOp(USERNAME is ${USERNAME})

exten = _X.,8,NoOp(username is ${username})


the REALTIME function returns a delimited string, it doesn't automatically
assign variables...


Perhaps you are thinking of the realtime application which would assign
variables? That has however was deprecated in 1.4 and has been removed from
1.6 in favour of the function which only returns a string...


d

On 29 March 2010 21:42, Jason Walker jason.wal...@amgsrv.com wrote:

  It seems that my realtime is not assigning channel variables correctly.



 INFO

 Asterisk 1.6.0.26



 Exten.conf

 exten = _X.,1,NoOp()

 exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})

 exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})

 exten = _X.,4,NoOp(DEVICE is ${DEVICE})

 exten = _X.,5,NoOp(USERNAME is ${USERNAME})

 exten = _X.,6,NoOp(username is ${username})





 CLI



 -- Executing [...@default:1] NoOp(SIP/1156-55ce, ) in new stack

 -- Executing [...@default:2] Set(SIP/1156-55ce, DEVICE=SIP/1156)
 in new stack

 -- Executing [...@default:3] Set(SIP/1156-55ce,
 NULL=username=john.smith,name=John
 Smith,department=Dept_A,routable=no,extension=1234,device=SIP/1156,voicemail=no,monitor=yes,visible=yes,date_modified=2010-02-09
 14:12:01,) in new stack

 -- Executing [...@default:4] NoOp(SIP/1156-55ce, DEVICE is
 SIP/1156) in new stack

 -- Executing [...@default:5] NoOp(SIP/1156-55ce, USERNAME is )
 in new stack

 -- Executing [...@default:6] NoOp(SIP/1156-55ce, username is )
 in new stack



 So I can see it is getting info from the database in Line 3



 But only the direct set variable command (Line 2) and Result (Line 4) work



 Lines 5 and 6 do not get the john.smith assigned



 Help





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Re: [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid

[D Tucny]

On 26 February 2010 12:38, Trevor Peirce tpei...@digitalcon.ca wrote:

 Charles Wang wrote:
  The sip.conf of MYE1 likes below:
  [MYPBX]
  type=peer
  host=mypbx.abc.com http://mypbx.abc.com
  nat=no
  disallow=all
  allow=g729
  canreinvite=yes
  qualify=no
  context=default
  insecure=port,invite

 Add sendrpid=yes here.

  The sip.conf of MYPBX likes below:
  [MYE1]
  type=peer
  host=mye1.abc.com http://mye1.abc.com
  nat=no
  disallow=all
  allow=g729
  canreinvite=yes
  qualify=no
  context=did
  insecure=port,invite

 Add trustrpid=yes here.

 A. Why can't I receive the CALLERID from MYPBX(the secondary
  server)? I am sure I use Set(CALLERID(num) for it.
 
 B. Why does the CALLERID that sends from MYE1 become
  as Anonymous? How can I fix it with the correct orginal
  callerid(912345678)?
 
 C. Why does my FROM message become as Anonymous
  sip:anonym...@anonymous.invalid instead of  912345...@mye1.abc.com
  mailto:912345...@mye1.abc.com ?

 You see this because, even though the number has been made available to
 you, it's marked as a blocked call.  Your server is honoring this and
 blocking the number when it dials the next server.  By using Remote
 Party ID, you'll be able to carry this information forward to your next
 server.


Or you could override the presentation using CallerPres function (in 1.6) or
SetCallerPres dialplan command in previous versions...

e.g. Set(CallerPres()=allowed) before doing the Dial to MYPBX.

d
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Re: [asterisk-users] Redirect call based on CLI???

[D Tucny]

Or, alternatively using the 's' priority...

exten = 845,1,Verbose(3, Incoming call from ${CALLERID(all)}) ; this will
be priority 1
exten = 845/12345678,n,Goto(blacklist) ; the n will make this priority 2
exten = 845/23456789,s,Goto(blacklist) ; the s will make this also priority
2
exten = 845/34567890,s,Goto(blacklist) ; again, priority 2
exten = 845/09876543,s,Goto(whitelist) ; and again
exten = 845/98765432,s,Goto(whitelist) ; and again
exten = 845/87654321,s,Goto(whitelist) ; and again
exten = 845,s,Verbose(3, CLI (${CALLERID(num)}) is neither blacklisted or
whitelisted) ; last s, so last priority 2, this time with no pattern
exten = 845,n,Dial(SIP/somewhere,180) ; this will be priority 3
exten = 845,n,Hangup() ; priority 4
exten = 845,n(blacklist),Verbose(3, CLI (${CALLERID(num)}) is blacklisted)
; priority 5
exten = 845,n,Hangup(21) ; priority 6, cause code 21 = rejected
exten = 845,n(whitelist),Verbose(3, CLI (${CALLERID(num)}) is whitelisted)
; priority 7
exten = 845,n,Dial(SIP/somewhereelse,180) ; priority 8
exten = 845,n,Hangup() ; priority 9

You can see the priorities from the asterisk cli by doing a 'dialplan show
context'

d

On 26 February 2010 01:33, Mark Hulber asterisk.ad...@hulber.com wrote:

 Since you are using 'n' notation, you might not have your statements
 aligned.  You can label your statements as below:

 exten = s,n,Answer
 exten = s,n,GotoIf($[${CALLERID(name)} != UNAVAILABLE]?ans)
 exten = s,n,Set(CALLERID(name)=${CALLERID(number)})
 exten = s,n(ans),NoOp


 ; Banned
 exten = s/708857500X,ans+1,Goto(banned,1)  ;
 exten = s/9044898017,ans+1,Goto(banned,1)  ;
 exten = s/8883222785,ans+1,Goto(banned,1)  ;

 ; Specifically routed

 exten = s/9165553456,ans+1,Goto(markivr,1)  ; Allowed
 exten = s/19165553456,ans+1,Goto(markivr,1)  ; Allowed

 ; Default
 exten = s,ans+1,Goto(mainmenu,s,1)

 On 2/25/2010 10:11 AM, Brian wrote:
  On Thu, 2010-02-25 at 03:00 -0800, Kyle Kienapfel wrote:
 
 
 http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
  Has example
  exten =  s,1,Answer
  exten =  s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD)
  exten =  s,2,Set(CALLERID(name)=Good Person)
  exten =  s,3,Dial(SIP/goodperson)
 
  for white list
 
  exten =  s/123123123,1,Dial(SIP/phoneA)
  exten =  s/456456456,1,Dial(SIP/phoneA)
  exten =  s,1,Dial(SIP/phoneB)
 
 
 
  Thanks Kyle.
 
  I tried the example given but I could not get this to work - basically
  if I dial it from any phone that does not match 0800800800 (for
  illustration) it hangs up the channel with an error.
 
  exten =  845/0800800800,n,Set(CALLERID(name)=EVIL BASTARD)
   Auto fallthrough, channel 'SIP/1000-0017' status is 'UNKNOWN'
 
  I'm struggling to work out the logic here of a non-match, but this was
  not caught by i or s in error, so I'm probably missing some brain
  connection here.
 
  However, I've managed to do what I want using gotoif statements matching
  caller id - but I'd be interested to work out how the above is meant to
  branch on a non-match.
 
 
 

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Re: [asterisk-users] problem in upgrading to 1.6.1.0

[D Tucny]

In the 1.6.1.* branch the line type=peer seems to be required on each
user...

d

2009/9/19 Örn Arnarson o...@arnarson.net

 Sorry I wasn't more specific.

 The error message is just the standard 'Can't find that extension'.

 The problem is, however, that asterisk parses users.conf (and doesn't
 complain), but none of the users specified therein are loaded into the
 dialplan or even shown as peers (using sip show peers/users). A
 downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan.

 Regards,
 Örn

 2009/9/18 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
 :
   Örn Arnarson o...@arnarson.net writes:
 
  I'm seeing the same behavior in 1.6.1.6.
 
  Any info on this?
 
  It would be helpful if you copied the exact error message involving the
  username field.
 
 
  /Benny
 
 

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Re: [asterisk-users] Call getting stucked !!

[D Tucny]

Check your VICIdial logs and try to debug the VICIdial side of things... It
could be something along the lines of agent hits hangup, web interface goes
to add hangup command into the manager queue, fails due to a lock on the
table so call stays up... This wouldn't be an asterisk issue...

You need to a) calm down b) realise you are asking people to give you help
for free c) learn how to ask good questions, in the right place d) learn how
to diagnose problems and give detailed accurate information regarding those
problems
d
2009/9/12 David @ULC ucoms2...@gmail.com

 I am GETTING tired of CALLS getting stucked !!!

 Any help ?


 On Thu, Sep 10, 2009 at 2:52 AM, David @ULC ucoms2...@gmail.com wrote:


 8186223080 : No entry in the log file !!!



 On Thu, Sep 10, 2009 at 2:06 AM, David @ULC ucoms2...@gmail.com wrote:


 Local/718186223...@d 718186223...@default Up
 Dial(SIP/18186223...@sip209||t


 I see this in my Asterisk when I do

 show channels




 On Thu, Sep 10, 2009 at 1:49 AM, David @ULC ucoms2...@gmail.com wrote:


 I don't know where is the problem. May be with VOIPSwitch OR may be with
 Asterisk..

 Call getting stuck : My agent hang up the call but in Active calls , I
 see call connected and getting charged

 I use VOIP and NOT PSTN

 Didnt check the Asterisk CLI. Can I get any history of what asterisk
 REALLY had ?




 On Wed, Sep 9, 2009 at 11:41 PM, David @ULC ucoms2...@gmail.comwrote:

 I am using asterisk.

 I also have an access to VOIPSwitch ver 2 where I can see live calls.

 Many times I have seen that my calls are getting strucked and then it
 gets disconneected after 59 mins ( as settings are done accordingly in
 VOIPSwitch)

 What could be the reason ?






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Re: [asterisk-users] Meaning of requested special control 20, passing it to SIP

[D Tucny]

2009/8/15 John Novack jnov...@stromberg-carlson.org

 Received this on the console
 -- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it
 to SIP/magicjack-08225a58


 Did a Google search, but reached a dead end
 Can anyone explain?
 Something need to be changed in my configuration?
 The call completed satisfactorily.

 Inbound IAX trunk - outbound to SIP provider magicjack ( no dongle )

 Asterisk 1.4.26


Special control 20 is defined as AST_CONTROL_SRCUPDATE = 20, /*!
Indicate source of media has changed */

When two channels are bridged they'll both be sent this control to indicate
that audio is now being sent from the other channel...

d
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Re: [asterisk-users] Cisco 7960 Multiline phone

[D Tucny]

With that phone what you really probably want to do is just configure them
all with the same details...

i.e.

# Line 1 appearance
line1_name: incoming
line1_shortname: Incoming (Line1)
line1_authname: incoming
line1_password: password

# Line 2 appearance
line2_name: incoming
line2_shortname: Incoming (Line2)
line2_authname: incoming
line2_password: password

# Line 3 appearance
line3_name: incoming
line3_shortname: incoming (Line3)
line3_authname: incoming
line3_password: password

# Line 4 appearance
line4_name: incoming
line4_shortname: Incoming (Line4)
line4_authname: incoming
line4_password: password

# Line 5 appearance
line5_name: incoming
line5_shortname: incoming (Line5)
line5_authname: incoming
line5_password: password

# Line 6 appearance
line5_name: 102
line5_shortname: Ext. 102 (Line1)
line5_authname: 102
line5_password: password

in the phone config file...

Then, in extensions.conf

exten = workhours,1,Dial(SIP/incoming)
exten = workhours,n,Voicemail(100,u)
...

The phone will only actually register multiple times for 'incoming' though
asterisk just handles that and calls to 'incoming' will come through on the
lowest available line and show as call waiting with an 'Answer' soft key
allowing the next call to be answered placing the current call on hold...

Seems to be exactly what you want...

d


2009/8/12 Jimmy Ezell jez...@hmhca.com

  Sorry for not being real clear.

 What I have is 1 front desk phone only with 6 lines
 Front Desk Phone line 1 - incoming extension 1
 Front Desk Phone line 2 - incoming extension 2
 Front Desk Phone line 3 - incoming extension 3
 Front Desk Phone line 4 - incoming extension 4
 Front Desk Phone line 5 - incoming extension 5
 Front Desk Phone line 6 - inside office extension

 If incoming line 1 is busy I want the next incoming call to come in on line
 2.
 If incoming line 2 and 3 are busy but 1 is free the next call should got to
 line 1.

 So lines 1 and 2 might get a lot of calls but only on really busy days will
 calls make it up to lines 4 and 5.

 Does that make sense?  Anyone have the solution?


 *Jimmy Ezell
 *


  --
 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons
 *Sent:* Tuesday, August 11, 2009 12:39 PM

 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone

  Jimmy,



 To clarify, you want to configure the phones like this where p means phone
 and l means logical line:



 Phone 1:

 P1l1

 P1l2

 P1l3



 Phone 2:

 P2l1

 P2l2

 P2l3



 Phone 3:

 P3l1

 P3l2

 P3l3



 It sounds like (and looks like) you’re dialing all of the extensions on one
 phone at the same time, which is why they’re ringing and ringing. What you
 want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1)
 in the dial command to ring them simultaneously. asterisk will then fail
 through if none of the phones answer in time.



 -Dave



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jimmy Ezell
 *Sent:* Tuesday, August 11, 2009 3:05 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone



 Thanks for the help, I really appreciate the feedback.



 I tried ringing them all at the same time as you suggested:

 exten =
 workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomming4SIP/incomming5)

 but it does very strange stuff:

 - I have to push the extension button twice to answer.

 - More then one extension shows off hook at the same time (Maybe 2 or 3 of
 the 5 will show off hook on the phone)

 - When I hang up the phone starts to ring again even though there is no
 caller



 I tried ringing them in order:
 exten = workhours,1,Dial(SIP/incomming1,5,r)
 exten = workhours,n,Dial(SIP/incomming2,5,r)
 exten = workhours,n,Dial(SIP/incomming3,5,r)
 exten = workhours,n,Dial(SIP/incomming4,5,r)
 exten = workhours,n,Dial(SIP/incomming5,5,r)

 exten = workhours,n,Macro(voicemail,100)



 Now I see the call march along each of the extensions until it gets to the
 end goes to voice mail.



 What I really want is for the call to go to only one of the unused lines
 and then fall straight through to voicemail after the timeout.

 Anyone have some thoughts on getting it to work that way?



  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons
 *Sent:* Tuesday, August 11, 2009 10:05 AM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone

 Yes each extension needs to be configured separately in the cisco CNF file.



 I use a distinct extension on each phone (2 phones can’t register to one
 ‘extension’ afaik) and ring them in order:



 1,1,Dial(SIP/xx)

 

Re: [asterisk-users] 7940g

[D Tucny]

2009/8/11 Chuck Coleman p...@2cci.com

  I have 6 Cisco 7940g phones and I would like to add them to my Asterisk
 2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco’s to
 register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to *
 no* but it still does not register. Please advise.


It just works... unless there's something very wrong with your config... let
us see your config... Erm, * 2.6.2?

d
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Re: [asterisk-users] Going to VM after 180 seconds in queue

[D Tucny]

2009/8/8 Dan Pilcheck pilch...@gmail.com

 Hello all,

 This is a VICIDial server and I am looking to send calls to VM box
 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
 This would be inserted between exten = s,8,Background(open) and exten
 = s,9,AGI.
 From what voip-info has [not] told me, the AGI doesn't allow for a
 timeout to be set.
 I'm hoping to find an option along the lines of the Dial() ringtime,
 but no luck.
 Gosub() looked interesting, but I don't think quite fits my needs either

 Could someone please offer a little insight on this situation and
 point me towards the right command to be playing with?


How about just setting the hold time in vicidial to 3 minutes, specifying a
drop action of extension and specifying drop extension of 2100?

d
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Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message

[D Tucny]

Perhaps it's only basic in certain parts of the world... I know I've never
experienced a voicemail system with such a feature...

I'm not saying having the option would be bad... but... I'd prefer voicemail
to get some more common, more requested of me, features first and that's
personally where I'd spend my effort... But... feel free to either put a
patch together yourself or offer a bounty...

d

2009/8/1 Emrah e...@ekanet.net

 Dom,

 Quoting myself from your original message:
  I know everything is possible to be done via the Dialplan. I could just
  have a Playback to achieve this or pretty easily code my own voicemail
  app via AGI too.

 I am not asking how to do this. I know everything is possible with
 Asterisk.
 In my opinion, the feature I am talking about is a very basic feature of
 Voicemail systems and I think it should be natively implemented in
 Asterisk.

 Thanks for your hint though. :)
 Emrah

 Don Kelly wrote:
  How 'bout setting up an extension that simply plays an announcement and
  hangs up. Then transfer calls from extensions that don't want messages to
  this extension.
 
  You could have a few extensions with a few different recordings to suit
  different situations.
 
  --Don
 
  Don Kelly
 
  PCF Corp
  People Come First
  651 842-1000
  888 Don Kell(y)
  651 842-1001 fax
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emrah
  Sent: Friday, July 31, 2009 11:37 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Voicemail feature: enable or disable the
  ability to leave a message
 
  Mark,
 
  I think you did not understand my message.
  I am accustomed to have the option to allow or disallow the recording of
  a message in my voicemail, even my mobile carrier provides it. E.g.: I
  can record a message that says Please call back later, I am currently
  on the phone. without any beep tone or the possibility to leave a
 message.
  I know everything is possible to be done via the Dialplan. I could just
  have a Playback to achieve this or pretty easily code my own voicemail
  app via AGI too.
  I don't catch what you are trying to tell about reading the input in the
  Dialplan...
  My scenario is pretty simple, there should be an option in the voicemail
  to allow the user to choose whether he accepts messages or not.
 
  Cheers!
  Emrah
  Mark Michelson wrote:
 
  Emrah wrote:
 
 
  Hi,
 
  I think there is an essential option of the Voicemail application that
  is missing.
  I would like to suggest the implementation of a function to give the
  user the ability to either allow or disallow the recording of messages.
  If the ability to record a message is disabled, options u, s, and b
 must
  not be considered in order to avoid the playback of messages such as
  Please leave your message after the tone...
 
 
  the usecase is simple. A person could record a greeting that says
 please
  callback later instead of asking to leave a message. usefull also to
  record afterhour messages.
 
  What do you think?
 
  Regards,
  Emrah
 
 
 
  There is no reason to place this logic in the Voicemail application
 
  itself. If
 
  you wish to give users the option of leaving a voicemail, it can easily
 be
 
  done
 
  in the dialplan by playing prompts and reading the input of the user.
 
  Then,
 
  based on the input, you can choose whether to run the Voicemail
 
  application.
 
  Mark Michelson
 
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Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?

[D Tucny]

2009/8/4 Faheem faheem_...@yahoo.com

 how to implement CLONED LINE Feature in asterisk

 Hey, I want to implement Clone Line feature in asterisk. I am using
 SPA-2100.
 The feature should work in this way.

 There are two ports in the SPA-2100 both are registered with asterisk with
 same username/password, and have the same (phone number)



   *No one on the phone *

 *One phone in use *

 *Both phones in use *

 *Incoming Calls *

 Both phones ring

 Phone in use receives call waiting notification, unused phone rings

 Both phones receive call waiting notification

 *Outgoing Calls *

 Both phones can call out

 The unused phone can call out

 Neither phone can call out


  * Inbound:
   - Both ports will ring. Whichever port is picked up first, will field
 the call.
   - Any additional calls that come in would give call waiting
 notification to the first line, and ring the second line.
   - Once the second line is being utilized, all incoming calls will be
 notifications in the form of call waiting beeps.

  * Outbound:
   - You will have the ability to dial out from port one.
   - You will be able to dial a different party on port two.

 *** Note ***
  - If you have an active call on port one, and pick up port two,
 you will NOT have the same call that is currently active on port one. The
 Cloned Line will share the same voice mail and will have the same telephone
 number as the original telephone line.

   -  The Cloned Line is NOT a second telephone number.  The telephone
 number that is assigned to the second phone port on the device is the same
 telephone number as the number assigned to phone port one.


In sip.conf
[line1]
username=line1
secret=line1password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default

[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default

In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()

or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
exten = 1234,n,Hangup()

then stdexten would be default as comes in the sample configs...

That should be everything you want if you configure the SPA-2100 to register
line 1 with username line1 and line 2 with username line2...

d
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Re: [asterisk-users] Message Waiting Indicator on DAHDI line

[D Tucny]

2009/8/5 Mike asterisk-us...@norgie.net

 On Tue, Aug 04, 2009 at 03:35:22PM -0500, Doug Bailey wrote:
 
  This code is designed to handle Message Waiting Indication (MWI) incoming
 on FXO
  line.  This data could very well be embedded in your CID spill as part of
 an
  MDMF message that also contains the caller id information.  (See
 main/callerid.c
  in the callerid_feed function.)
 
  If your incoming line has a mailbox associated with it, the MWI
 information will
  be pushed to that mailbox.  You may want to look at how your mailboxes
 are defined
  and the channels to which they are associated.
 
  Doug

 Doug,

 Thanks, that pointed me in the right direction.  I found the problem in
 chan_dadhi.conf, I just don't totally understand it.

 signalling=fxo_ks
 echocancel=yes
 pulsedial=yes
 ;group=1
 channel=1
 ;mailbox=3203

 signalling=fxs_ks
 ;group=1
 usecallerid=no
 faxdetect=none
 signalling=fxs_ks
 rxgain=4
 txgain=4
 ;callerid=1234
 channel=3

 signalling=fxs_ks
 ;callerid=1234
 echocancel=yes
 ;group=2
 channel=4

 Forgive the messy config, it's been a bit butchered in my efforts to get
 it working.  The MWI problem was solved when I commented out the
 mailbox=3203 for chan 1.  This is a phone that is dailed, along with the
 SIP phone, when the FXO lines ring.  The SIP phone is tied against the
 same mailbox in the sip.conf.

 What I don't understand is, I thought the mailbox= in chan_dahdi.conf
 was there to do a stutter dialtone on the FXS port when there was a
 message in that mailbox?  Why should commenting it out help?

 I assume that what happens is that the FXO line rings, so Asterisk rings
 the FXS phone as per the extensions.conf, this creates a MWI event which
 goes to the voicemail system, which then passes a MWI event to the SIP
 phone (as per sip.conf)?  Or I could just be talk rubbish!


The problem is that your mailbox line was below channel=1, as such, it
applied to the next channel, channel=3 not channel=1...

d
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Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?

[D Tucny]

I'd suggest using different user names and getting asterisk to handle the
cleverness... And, well, doing it this way is pretty simple, straight
forward, basic asterisk functionality...

Trying to get two different instances registered as the same user, is, as
you've found out, not going to be trivial to implement... It's just not how
it works...

If you implemented what I suggested below, using two different usernames for
the two ports on the SPA, it would just work...

d

2009/8/5 Faheem faheem_...@yahoo.com


 By placing OPENSIP in front of Asterisk, we can register multiple accounts,
 and we can successfully make call for Outgoing only. But in case of incoming
 it fails.

 If two users are registered with asterisk or OpenSIP then the user that is
 registered latest is considered to be valid, and he is able to make calls,
 other user with earlier registration can not make call.
 My point here is in chain_sip.c what are variables or structure that need
 to maintain so that we can consider all registered users as active users.

 Thanks!
 Faheem

 --- On *Wed, 8/5/09, D Tucny d...@tucny.com* wrote:


 From: D Tucny d...@tucny.com
 Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in
 asterisk?
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Wednesday, August 5, 2009, 11:06 AM


 2009/8/4 Faheem 
 faheem_...@yahoo.comhttp://mc/compose?to=faheem_...@yahoo.com
 

  how to implement CLONED LINE Feature in asterisk

 Hey, I want to implement Clone Line feature in asterisk. I am using
 SPA-2100.
 The feature should work in this way.

 There are two ports in the SPA-2100 both are registered with asterisk with
 same username/password, and have the same (phone number)



   *No one on the phone *

 *One phone in use *

 *Both phones in use *

 *Incoming Calls *

 Both phones ring

 Phone in use receives call waiting notification, unused phone rings

 Both phones receive call waiting notification

 *Outgoing Calls *

 Both phones can call out

 The unused phone can call out

 Neither phone can call out


  * Inbound:
   - Both ports will ring. Whichever port is picked up first, will
 field the call.
   - Any additional calls that come in would give call waiting
 notification to the first line, and ring the second line.
   - Once the second line is being utilized, all incoming calls will be
 notifications in the form of call waiting beeps.

  * Outbound:
   - You will have the ability to dial out from port one.
   - You will be able to dial a different party on port two.

 *** Note ***
  - If you have an active call on port one, and pick up port two,
 you will NOT have the same call that is currently active on port one. The
 Cloned Line will share the same voice mail and will have the same telephone
 number as the original telephone line.

   -  The Cloned Line is NOT a second telephone number.  The telephone
 number that is assigned to the second phone port on the device is the same
 telephone number as the number assigned to phone port one.


 In sip.conf
 [line1]
 username=line1
 secret=line1password
 type=friend
 host=dynamic
 context=outboundcalls
 mailbox=1...@default

 [line2]
 username=line2
 secret=line2password
 type=friend
 host=dynamic
 context=outboundcalls
 mailbox=1...@default

 In extensions.conf
 [default]
 exten = 1234,1,NoOp(About to dial both phones)
 exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
 exten = 1234,n,Hangup()

 or for trunk
 [default]
 exten = 1234,1,NoOp(About to dial both phones)
 exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
 exten = 1234,n,Hangup()

 then stdexten would be default as comes in the sample configs...

 That should be everything you want if you configure the SPA-2100 to
 register line 1 with username line1 and line 2 with username line2...

 d

 -Inline Attachment Follows-

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Re: [asterisk-users] asterisk 1.6 call forwarding

[D Tucny]

2009/7/31 pepesz76 pepes...@o2.pl

 Dear All,

 I'n trying to make a simple call forwarding, however I have small
 problem when evaluating an expresion.

 Here is my extensions.conf
 ...


 ; Unconditional Call Forward
 exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
 exten = _#21*X.,2,Hangup()
 exten = #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})})
 exten = #21#,2,Hangup()
 ...
 exten = 50,1,Set(CFIM=${DB(CFIM/${EXTEN})})
 exten = 50,n,GotoIf($[${CFIM}=]?start)  ;- THIS IS WRONG, but not
 sure what should it look like?
 exten = 50,n,Dial(SIP/${CFIM},30)
 exten = 50,n,Dial(SIP/${EXTEN},30
 ...

 First part properly sets and deletes string in database
 Second part works as the forwarding is set, however if it is not set
 then CFIM is empty and I got:
 WARNING[9752]: ast_expr2.fl:434 ast_yyerror: ast_yyerror():  syntax error:
 syntax error, unexpected '=', expecting $end; Input:
 =''

 Can someone suggest the solution?


What it's doing in this case is first extracting the variable value then
performing the evaluation of the condition, so, when there is no value set,
what it's effectivly trying in the condition is $[=], hence the unexpected
'=' message. If you quote the variable, i.e. $[${CFIM}=] then in the
event the variable is unset or contains an empty value the condition will
still be at least $[=] and it should work...

d
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Re: [asterisk-users] MeetMe Options Enter Leave Sound

[D Tucny]

2009/7/24 Stefan Schmidt s...@sil.at

 Hello,

 i´ve a question about the Meetme Options. How could i play a enter and
 leave sound but without recording the user name first. I just want a
 User joined conferenc and a user leaved.

 With the i or I Option i have to record the name first.

 Is there any way of doing this? As i can see in the Meetme help the
 background agi couldnt be used on non dahdi channel which i will have on
 this server cause there is no direct Pri link, just SIP.


By default join and leave sounds are played when someone joins and leaves
the conference...

d
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Re: [asterisk-users] Not getting inbound CallerID name on Asterisk

[D Tucny]

Looking at the code, asterisk doesn't know how to handle RPID in an INFO
message, so it just responds with an OK and goes on with it's business...
The fact that the message has the name of the called party, rather than the
calling party probably wouldn't help even if Asterisk did understand it... I
would have expected you to at least get 'pending' through as the name
though...

d

2009/7/31 Chris Douglas chris.doug...@pioneerballoon.com

 The following pastebin shows the inbound call, inbound INFO containing the
 Remote-Party-ID string, and the SIP acknowledgement of the INFO.  Asterisk
 does not send the data from the Remote-Party-ID string on to the phone, nor
 does it set the CALLERID(name) variable after receiving the message.

 http://pastebin.com/m45e0adbd

 Thanks,
 Chris

 On Sun, Jul 26, 2009 at 1:19 PM, Chris
 Douglaschris.douglas at pioneerballoon.com wrote:
  We have an inbound PRI connected to our Cisco 3825 router which is then
  passing the calls to Asterisk as SIP calls.  We're getting the CallerID
  number but not the CallerID name.  We are seeing the name in the RPID
 field
  with a SIP trace on the Asterisk box but don't understand why it's not
  registering as the CallerID name.

 What do you get when you enable debugging on the asterisk cli?

 core set verbose 3

 make a call. Do you see caller ID going through? What does your
 dialplan look like? You can use NoOp() calls to pop out values
 including a caller ID if it exists.

 -Original Message-
 From: Chris Douglas [mailto:chris.doug...@pioneerballoon.com]
 Sent: Sunday, July 26, 2009 12:20 PM
 To: 'asterisk-users@lists.digium.com'
 Subject: Not getting inbound CallerID name on Asterisk

 We have an inbound PRI connected to our Cisco 3825 router which is then
 passing the calls to Asterisk as SIP calls.  We're getting the CallerID
 number but not the CallerID name.  We are seeing the name in the RPID field
 with a SIP trace on the Asterisk box but don't understand why it's not
 registering as the CallerID name.

 Here is a link to pastebin with the Sip trace.  In it you can see the RPID
 is seen from the Asterisk box but it is not used/sent to the phones.

 http://pastebin.com/m45e0adbd

 Here is the section from Sip.conf describing the Cisco 3825 connection.  We
 have tried type as both friend and peer as it is now with no change.

 [cisco_3825]
 context=default
 type=peer
 host=10.0.0.10
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 trustrpid=yes
 sendrpid=no

 All phones are not receiving the CallerID name, here is a sample from
 sip.conf of a phone config.

 [8670]
 secret=8670
 context=ict_sip
 type=friend
 host=dynamic
 call-limit=5
 agentlogin=yes
 mailbox=8...@ictvm
 progressinband=no
 sendrpid=yes


 Any help is greatly appreciated!


 Thanks,
 Chris Douglas
 Technical Services Manager
 Pioneer Balloon Company




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Re: [asterisk-users] PHP AGI Problems

[D Tucny]

2009/5/27 Atlanticnynex atlanticny...@gmail.com

 (Accidentally posted this to asterisk-dev, should be here)

 fgets is only returning one character... either when run as an AGI or
 run as a test on PHP on CLI...
 Example, enter , then fgets returns '3'.


It's not... There are problems with the way you are handling the return
data...



 Also, GET DATA seems to be returning early and the loop keeps
 prompting 'invalid'...


That's happening because you are sending junk to asterisk...



 Any suggestions on how to improve my AGI class so it actually works?


Have a look at http://www.voip-info.org/wiki/view/Asterisk+AGI+php...



 Thanks.

 [code]

snip



 while ($error_count  5  !$msg_result) {
   $result = $AGI-send_cmd(GET DATA /bswitch/menu/enter-msg-id 3000 4);


   $msg_id = $result[result];

Problem here, you're looking to extract the value for key result (which as
it's not quoted could be interpreted as a constant) from the array $result,
but, $result isn't an array


   if ($msg_id  1000){
   echo \ndebug: msgid  1000, invalid ($msg_id)!\n;

Junk is being sent to asterisk which is responding with errors (echo sends
to STDOUT too), you're only getting one line at a time of response, so,
after the first time in you are not getting the results you'd expect in the
GET DATA, you're getting the rest of the error messages from the junk you
sent...


 ?php
 //AGI.class.php
 class AGI {
   public $agivars;


snip



   function send_cmd($cmd){
   fputs(STDOUT, $cmd .  \n);
   fflush(STDOUT);
   $data = fgets(STDIN, 4096);
   return $data;
   }

So, as said above, you're sending the command here, then taking the next
line from asterisk and dumping it back unprocessed in any way...

When you're sending the GET DATA command, if you enter , what you are
getting back is...

'200 result=\n'

$result actually contains this value, however, when you set $msg_id to
$result[result] what you get is the first character in $msg_id, so, in this
case you get '2'...

As this is lower than 1000 you go into the invalid number part of your code
where you echo a debug message then send the command to play the invalid
sound file... Sending the command to play the sound file results in a 1 line
response being received, your echo command put out two invalid lines, so,
what is actually received as a response this time is...

'510 Invalid or unknown command\n'

The next command you send is the GET DATA command to try and get a new
number, however, at this point there is already data waiting in the buffer,
so as soon as the data is sent, your fgets returns straight away with...

'510 Invalid or unknown command\n'

as a result of the second line of debug output that you sent...

$result contains this string and again, you get the first character in
$msg_id, so now $msg_id equals '5', this again is lower than 1000, so you go
into the invalid number part of your code and drop more junk into asterisk
and continue...

I hope that's useful in explaining the problems you are experiencing... as I
said above, check out the page on voip-info, it should be enough for you to
get this fixed up nicely... One think you'll notice about the examples on
there, debug output is sent to a file, that way you can record information
for debugging without polluting your agi...

d
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Re: [asterisk-users] Question with Asterisk and call waiting ${CALLERID(name/num)}

[D Tucny]

2009/4/30 Justin Piszcz jpis...@lucidpixels.com

 Hello,

 I am using an SPA3102, all is working with asterisk 1.4, voice mail,
 outbound calling etc, and it even passes the cid name/num to my analog
 phone.  However, when someone is calling me, I hear the beeps but the
 caller-id information is not showing up on my phone, is this an SPA3102
 problem or an asterisk configuration issue?


Make sure you have CWCID enabled on the SPA and that your phone supports
it...

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Re: [asterisk-users] Faxing and TIFF files

[D Tucny]

2009/4/22 Michael mich...@networkstuff.co.nz

 I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing.

 Does anyone know of a way, either while producing the file, or after, to
 tell
 how many pages have been produced? (without manually viewing the file)


tiffinfo? then count the number of data blocks...

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Re: [asterisk-users] run dialplan when open line

[D Tucny]

2009/4/22 michel freiha mich...@gmail.com

 Hi all,
 Does asterisk support the following scenario? I need when a customer who
 own an endpoint registered on asterisk open the line, the asterisk will run
 a specific AGI script inside the endpoint context?


You mean when they pick up the phone it'll automatically run the AGI?
If so, easy to do with dahdi channels using immediate=yes, the s extension
in that channels context would be executed on the phone going offhook, with
other channel types it depends on whether the endport device supports some
way of really automatically opening a channel... Most SIP devices will
generate their own dial tone and at a certain point, based on their local
dialplan, send the dialled string to asterisk to open a channel, if nothing
is dialled, they wouldn't contact the server... That said, their may be some
devices that would allow configuration such that they will automatically
dial a number when going offhook...

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Re: [asterisk-users] opening 2 and more channels on 1 SIP account

[D Tucny]

2009/4/18 Tamer Higazi th9...@googlemail.com

 Scenario:
 I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP
 cpu to take out the echo cancellation.

 Communication is done through the chan_capi interface module.

 If a call comes inside, and I forward it to the SIP account that is
 registered in the module, then all DECT phone do ring. But DECT / GAP
 phones are not designed for these issues.

 Scenario what a commercial PBX system does which has a ISDN board.

 Set up the phones:
 1 - queues through system messages the dect man station on which the
 cordless devices are registered to. the main station tells him the ID of
 the devices and I assign through the webinterface the numbers (DDI or
 MSN) to the devices.

 2 - set up is done!

 Call routine:


 Call in!
 1 - from the NT unit of my home line comes a call that goes to the PBX.
 2 - The PBX which receives the call extract the number (DDI or MSN) and
 compare it in the list of which phone it is (from step one)!
 3 - The PBX send a message queue to the base station to check if the
 phone is busy, if yes forget it. If no pass the call through. Done with
 sending a message to the base that the call is passed to this device,
 for that the other devices won't ring.

 Call out!
 1 - from the handset I make a call
 2- the PBX, sends a message to the base station asking who dialed the
 number.
 3 - the base station gives back the id, the outgoing number is set for
 that the call is passed through with the desired outgoing number.


 Now Asterisk, if SIP supports it receiving and placing several calls
 through one FXS port:

 the agi script:
 http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText

 1 - a call is placed
 2 - the agi script sends a message through the sip channel and the anser
 comes back, the answer is held in a variable
 3 - the variable had been worked out, and the MSN or DDI is set

 ---

 1  a call is received through the chan_capi interface
 2.  the dialplan knows which id belongs to the DDI or MSN number and
 calls the AGI script, which sends the message to the base station asking
 if the handset is available, busy or ready to receive calls.
 3. the script returns a value that is being worked out and the agi
 script is called again to tell the base station that the incoming call
 is for the handset id (let us say number 5), that not all phones do ring.
 4. the call is forwarded to the FXS port and that's it.


 This is how a usual PBX System in Germany and across europe do work. But
 if SIP or Asterisk do not support receiving and placing more calls
 through one FXS port and channel at the same time, then the DECT
 sollution can be dropped at all for me, and I shouldn't lose more time
 in this issue.


 DECT itself, is a well worked out technologie that gives you the chance
 to make a lot! It is programming work, not more then that.

 I hope all questions are being answered.


You are confused...

While DECT may well be capable of this sort of functionality and while
asterisk, and SIP are capable of this sort of functionality, you are using
an intermediate technology, a single POTS analogue connection, that isn't
capable...

You'll need a DECT base that either directly supports SIP for communicating
with Asterisk, or, with a more capable interface, such as ISDN, that allows
for more advanced communication and multiple channels...

The best you could probably hope to get using an FXS connection is that a
single inbound call could be routed to one of the handsets by using
distinctive ring, if the base supports it... However, you can not have more
than one call over one analogue FXS connection, this isn't an Asterisk or
SIP limitation, this is a limitation of the analogue connection...

Example SIP DECT devices:
http://www.snom.com/en/products/snom-m3-voip-phone/
http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-1814D885/03/hs.xsl/30395.htm
Multiple Siemens devices

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Re: [asterisk-users] opening 2 and more channels on 1 SIP account

[D Tucny]

2009/4/18 Tamer Higazi th9...@googlemail.com

 D Tucny schrieb:
  2009/4/18 Tamer Higazi th9...@googlemail.com
  mailto:th9...@googlemail.com
 
  Scenario:
  I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a
 DSP
  cpu to take out the echo cancellation.
 
  Communication is done through the chan_capi interface module.
 
  If a call comes inside, and I forward it to the SIP account that is
  registered in the module, then all DECT phone do ring. But DECT / GAP
  phones are not designed for these issues.
 
  Scenario what a commercial PBX system does which has a ISDN board.
 
  Set up the phones:
  1 - queues through system messages the dect man station on which the
  cordless devices are registered to. the main station tells him the
  ID of
  the devices and I assign through the webinterface the numbers (DDI or
  MSN) to the devices.
 
  2 - set up is done!
 
  Call routine:
 
 
  Call in!
  1 - from the NT unit of my home line comes a call that goes to the
  PBX.
  2 - The PBX which receives the call extract the number (DDI or
  MSN) and
  compare it in the list of which phone it is (from step one)!
  3 - The PBX send a message queue to the base station to check if the
  phone is busy, if yes forget it. If no pass the call through. Done
  with
  sending a message to the base that the call is passed to this device,
  for that the other devices won't ring.
 
  Call out!
  1 - from the handset I make a call
  2- the PBX, sends a message to the base station asking who dialed the
  number.
  3 - the base station gives back the id, the outgoing number is set
 for
  that the call is passed through with the desired outgoing number.
 
 
  Now Asterisk, if SIP supports it receiving and placing several calls
  through one FXS port:
 
  the agi script:
  http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText
 
  1 - a call is placed
  2 - the agi script sends a message through the sip channel and the
  anser
  comes back, the answer is held in a variable
  3 - the variable had been worked out, and the MSN or DDI is set
 
  ---
 
  1  a call is received through the chan_capi interface
  2.  the dialplan knows which id belongs to the DDI or MSN number and
  calls the AGI script, which sends the message to the base station
  asking
  if the handset is available, busy or ready to receive calls.
  3. the script returns a value that is being worked out and the agi
  script is called again to tell the base station that the incoming
 call
  is for the handset id (let us say number 5), that not all phones
  do ring.
  4. the call is forwarded to the FXS port and that's it.
 
 
  This is how a usual PBX System in Germany and across europe do
  work. But
  if SIP or Asterisk do not support receiving and placing more calls
  through one FXS port and channel at the same time, then the DECT
  sollution can be dropped at all for me, and I shouldn't lose more
 time
  in this issue.
 
 
  DECT itself, is a well worked out technologie that gives you the
  chance
  to make a lot! It is programming work, not more then that.
 
  I hope all questions are being answered.
 
 
  You are confused...
 no I am not






 
  While DECT may well be capable of this sort of functionality and while
  asterisk, and SIP are capable of this sort of functionality, you are
  using an intermediate technology, a single POTS analogue connection,
  that isn't capable...
 
 read the DECT specification from A-Z. In Germany we have digital (isdn
 analog adapter that do it). By the way, in Germany and many other
 European countries, BRI ISDN connections for household and companies are
 widespread.


ISDN is not analogue... you are confused... I know BRI ISDN connections are
widespread, I've used them extensively before... Yes, a BRI ISDN connection
can support two concurrent voice calls... Yes, a BRI ISDN connection can
pass all sorts of additional signalling information... but... you are
talking about a single analogue connection provided by an ATA device and a
DECT base that only has a single rj-11 analogue connection...

Show me the section of the DECT specifications where they give details on
how to run multiple analogue voice streams over a single standard analogue
circuit without special hardware and I'll admit defeat right now and go to
the pub to toast all the engineers involved in writing the specs as they
will have acheived something miraculous...



 commercial PBX systems are BRI ISDN and analogue FXS related. Call comes
 in, and call is going out.


And the same is true with asterisk, call comes in, processing happens, call
goes out... that is not what is at issue here... it's the basic
technological issue that the analogue interface will take a single call and
you seem to be hoping

Re: [asterisk-users] dialling multiple extensions in an internal context

[D Tucny]

2009/4/19 Matthew Pounsett m...@conundrum.com

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1


 Hi there.  I've done some googling around to try and find an example
 of what I'm trying to do, but it's one of those things that just seems
 hard to find the right terms to search for.  If there's some
 documentation out there on this, I'd appreciate being pointed in the
 right direction.   If not, then if someone has some ideas I'd
 appreciate that too. :)


 What I'm trying to do is set up some 'special' extensions in my
 internal context to change variables, or change something else in the
 session before dialling.  To be clearer, here's an example.  Say I've
 got this rather simple dial plan:

 [globals]
 TRUNK1=Zap/1
 TRUNK2=Zap/2
 TRUNK=${TRUNK1}

 [internal]
 _NXX,1,Dial(${TRUNK}/${EXTEN})


 I'd like to add an extension which I can dial before placing the
 actual call to change which trunk I'm using, like so:

 *55,1,SetVar(TRUNK=${TRUNK2})

 The problem is that once that's done, asterisk stops looking for me to
 dial an extension, and I'm trying to figure out how to get back to the
 top of the context and have asterisk wait for a new extension to
 dial.  I've dug through the Asterisk O'Reilly book, and googled around
 some, but haven't come up with the answer.

 Thoughts anyone?


How about, something like one of these...

exten = _*55NXX,1,SetVar(TRUNK=${TRUNK2})
exten = _*55NXX,n,Dial(${TRUNK}/${EXTEN:3})

or

exten = _*55NXX,1,SetVar(TRUNK=${TRUNK2})
exten = _*55NXX,n,Goto(internal,${EXTEN:3},1)

or

exten = *55,1,SetVar(TRUNK=${TRUNK2})
exten = *55,n,DISA(no-password,internal)

or

exten = *55,1,SetVar(TRUNK=${TRUNK2})
exten = *55,n,WaitExten(5)

or use your imagination :)

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Re: [asterisk-users] dialling multiple extensions in an internal context

[D Tucny]

2009/4/19 Matthew Pounsett m...@conundrum.com

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1


 On 18-Apr-2009, at 17:17, John F. Ervin wrote:

  Can't you handle that by defining an outbound route? set it to hit a
  trunk or set of trunks when the correct dial pattern is detected?

 Probably, for the one specific example of using a different trunk, but
 that doesn't solve the problem I asked about:  using a dial pattern to
 change variables or other session state then returning to waiting for
 an extension to be dialled.   Changing the TRUNK global is just one
 thing that could be done with it.  Other example use cases might be
 changing outbound caller ID, setting up call recording, etc.

 The WaitExten suggestion was pretty close.. it does what I want, but
 without giving a dial tone.  I ended up using DISA instead.


If you want to use WaitExten, but want a dialtone, that's possible too, just
need to insert a Playtones(dial) before it...

d
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Re: [asterisk-users] 2B Channel Transfer on XO-based T1

[D Tucny]

2009/4/15 Olivier oza-4...@myamail.com

 May I ask : beside saving channels, what are the benefits of TBCT over
 bridging calls inside Asterisk ?


I'm not aware of anything apart from saving channels...



 What about caller ids ? I would say caller id should passed over to final
 callee after bridging but that should need some kind of signaling update.


Passing the original callerID during a forward/unattended transfer is
standard North American functionality... Call comes in, forwarded out either
over the same, or alternate, trunk, callerid left intact... A feature that
can make using certain commercial PBXs in other countries somewhat
problematic...

Updating the caller information during a call, including feeding back
connected callee information, is I believe, a qsig/dpnss addition used
within a private network (though there may be instances that it can work in
a public network that I've not seen)...

d
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Re: [asterisk-users] duration of rfc2833 generated dtmf

[D Tucny]

2009/4/15 John covici cov...@ccs.covici.com

 Its not there and the link you gave me says its for sip originating
 rather than calls to a sip channel.

 on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
   It's been around awhile.  I've used it in 1.4  Check out this link for
   basic info:
 http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode
  
   John covici wrote:
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
   
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com)
 wrote
   One thing you might try is researching the SipDtmfMode command.
  It
  allows you to change the DTMF mode on an active channel.  A
 suggestion
  might be to set up the dial command with the M() option that point
 to a
  Macro that changes the DTMF to INBAND once you are connected to the
  problem number.  At least in theory, if your provider is expecting
  RFC2833 and they get inband, they should just ignore the inband
  signaling and pass it on as part of the audio stream.  The only
 problem
  is that this may only work if you use uLaw or aLaw for your codec
 and I
  don't know exactly how to set the tone duration without having a
  zapata.conf or dahdi.conf entry.  Even with one of those files, I
 don't
  know how Asterisk chooses to do the rfc2833 to inband translation
 or
  where it pulls the toneduration setting from if no PSTN interface
 is
  involved in the call.


It's been there from at least 1.0...

But, you are correct, it's for use on incoming SIP calls rather than
outgoing SIP calls...

d
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Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)

[D Tucny]

2009/4/14 jonas kellens jonas.kell...@telenet.be


 [r...@asterisk asterisk]# netstat -a -n -p | grep 5060
 udp0  0 0.0.0.0:50600.0.0.0:*
 3047/asterisk


 [r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
 listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes
 23:04:59.522498 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:01.233460 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:23.521076 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:24.520486 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:25.232068 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:26.231229 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:26.520308 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:28.231050 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:30.519957 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:32.230693 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:34.521843 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 925
 23:05:34.530587 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 530
 23:05:35.519255 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 925
 23:05:36.230336 IP 192.168.4.112.sip  192.168.4.248.sip: SIP, length: 540
 23:05:37.519077 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 925
 23:05:41.518720 IP 192.168.4.114.sip  192.168.4.248.sip: SIP, length: 925

 Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3047)
 Verbosity is at least 3
 asterisk*CLI sip debug
 SIP Debugging re-enabled
 asterisk*CLI

 and it stays that way...


So far, what you've provided as information suggests that asterisk is
running, that it is listening on the SIP port, udp 5060, that the phones are
sending information to asterisk, the machine asterisk is running on can see
the packets in a tcpdump, but, asterisk never sees what the phones are
sending going back the lack of anything showing up in the asterisk cli...

Make sure you set at least udp port 5060 in your iptables configuration as
ACCEPTed...

To do it as a one off, you can do 'iptables -I INPUT -i eth1 -p udp -m udp
-s 192.168.4.0/24 --dport 5060 -j ACCEPT', though unless you change your
configuration this will not survive across a reboot...

This wouldn't necessarily be enough to get SIP working, but, the phones
should at least register and you should be able to see them at least try to
make calls... If you get that far, you'll also need to open a range of ports
for RTP...

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Re: [asterisk-users] Siemens Gigaset Phones get mute function.

[D Tucny]

2009/4/8 Olivier oza-4...@myamail.com


 2009/4/8 Alan Lord (News) alansli...@gmail.com


 * DHCP Option 114 implemented.*
 * DHCP Option 120 implemented.*
 http://lists.digium.com/mailman/listinfo/asterisk-users

 What does it imply ?
 Provisionning from DHCP server ?


114 is for passing a URL to be displayed after boot
120 is for passing SIP servers (RFC3361)

So, I guess that would be a yes...

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Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs

[D Tucny]

2009/3/28 Jason Parker jpar...@digium.com

 D Tucny wrote:
  2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com
 
 
  Hi, Axel.
 
  Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later into rpmrepo.org
  http://rpmrepo.org)? That way we won't
have further fragmentation and a larger user base to test bits
 (which
will be distributed in stable, testing etc repos).
 
  Of course I'd love to contribute my changes to ATrpms.  Some of the
  small changes I made, such as adding OSLEC to the DAHDI RPMs, might
 be
  nice for ATrpms users.  I'll whip up some patches against the ATrpms
  sources.
 
  My problem with ATrpms, though, is that the RPMs make use of many
 custom
  macros that make them unbuildable outside the ATrpms environment.  I
  understand that might be necessary for RPMs like DAHDI that build
 kernel
  modules for several versions of several distros, where vanilla
 specfile
  code would get hairy.  (I think we had this discussion a couple of
 years
  ago on the ATrpms ML.)  Since I don't have to worry about multiple
  versions of multiple distros in my environment, I prefer to use
 vanilla
  specfile that will rebuild on anyone's CentOS 5 system.
 
 
  Alternatively, there's also the RPMS at
  http://packages.asterisk.org/centos/ which seem to have a nice spread of
  options available, including 1.4/1.6 packages, are pretty nicely
  modularised and seem to be kept pretty fresh... They do however seem to
  have some issues that your RPMS (and Axel's) don't (e.g. why wouldn't an
  init file be included? and where's the changelog?)... Perhaps it would
  be useful to help the digium packager build some better packages... That
  would also help with reducing fragmentation, if there were decent
  quality 'official' packages available then it would save the time and
  effort Axel and the rpmrepo.org http://rpmrepo.org folks too as they
  could in theory base any extras on those packages rather than needing to
  maintain the entire set...
 
  d
 

 As the author of the RPMs at http://packages.asterisk.org/ (as well as
 http://packages.digium.com/), and the maintainer of the repositories, I
 wanted
 to respond to this.

 I would love it if some of this were to happen.  I am very familiar with
 Axel
 and ATrpms - he has proven countless times that he knows what he's doing
 when it
 comes to this sort of thing.  Getting help/advice from somebody like him
 would
 be extremely beneficial.  As far as basing the ATrpms (or others) packages
 on
 the AsteriskNOW packages, if that is something that Axel (or others) wanted
 to
 do, I would be more than willing to help with whatever is needed.  On a
 somewhat
 related, and very interesting note - I found out yesterday that the latest
 trixbox beta is using these RPMs (without even needing to rebuild them, in
 some
 cases).  Hopefully that means I'm doing something right.

 D, the two issues you brought up are valid.  For the Asterisk RPMs, I
 honestly
 don't know why there isn't an init script - I actually thought there was
 one.
 FreePBX is what starts Asterisk in AsteriskNOW, so it was easily
 overlooked.  It
 will be there in future builds.


Cool... I tried the 1.6 packages out a couple of days ago, not using
FreePBX, the biggest issue I've found so far was the lack of init scripts,
both in the asterisk packages and the dahdi... I'll have a look at patching
the spec's to make them more usable without freepbx and feed back to you...


 As far as the changelog, it was one of those
 things that I intentionally left out for a while, and I kept meaning to do
 it
 later.  Really, it's because I'm not sure what should go into an RPM
 changelog
 (I'd love to hear from anybody that has any insight into that).


What would typically go in an RPM changelog (as with most changelogs) is a
note associated with each revision...

e.g. for the asterisk16 RPM you may have something like this (which I
knocked up from looking at the specs for each release) that makes it simple
to see why there is a new revision without having to dig through the
SRPMs...

%changelog

* Sat Mar 21 2009 Jason Parker jpar...@digium.com - 1.6.0.6-2
- specify asterisk:asterisk ownership and specific permissions of
directories and files that should be writable at runtime

* Tue Feb 24 2009 Jason Parker jpar...@digium.com - 1.6.0.6-1
- Update to 1.6.0.6
- add astapi variable and add provides with that variable for each child
package
- include func_audiohookinherit.so, aelparse and conf2ael
- change useradd to use '-r' to create a system account with UID below
UID_MIN and produce no output

* Wed Feb 11 2009 Jason Parker jpar...@digium.com - 1.6.0.5-1
- Update to 1.6.0.5
- add distname and distver variables to allow logic for building on other
dists
- add logic to modify buildreqs and reqs if building for suse/sles
- make newt buildreq

Re: [asterisk-users] Early Media

[D Tucny]

I can't say it's always been like this, as I don't recall, but, Background
in 1.0 behaved like this, answering the channel if it wasn't already
answered and playing the sound file/s until they finished or an exten was
dialed...

in 1.0 the 'skip' option would cause playback to be skipped if the channel
was not 'up', the 'noanswer' option would cause the channel to not be
answered
in 1.2 the options became 's' for skip and 'n' for noanswer though the
original 'skip' and 'noanswer' options are still valid even in 1.6

That said, in this example, you'd never leave background as it would sit
there playing the background_song file waiting for digits to be dialled...
using the dial option would be the way...

d

2009/3/27 Danny Nicholas da...@debsinc.com

 Is this correct for all versions, or does it start at 1.4 or 1.6?  I did
 put
 a YMMV on the comment, so my answer was not to be taken as fact.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
 Sent: Thursday, March 26, 2009 1:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Early Media

 On Wed, 2009-03-25 at 08:34 -0500, Danny Nicholas wrote:
  YMMV, but you might try this
 
  Exten = s,1,background(background_song)
 
  Exten = s,n,Answer() ;start billing

 This is not correct.  Background() automatically answers the call if it
 hasn't been answered already.

 The way to accomplish the task the original poster asked is to use the
 m option to the Dial() application.

 --
 Jared Smith
 Training Manager
 Digium, Inc.


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Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs

[D Tucny]

2009/3/26 John Morris aster...@zultron.com

 Hi, Axel.

 Axel Thimm wrote:
   How about merging in your changes/improvements/new packages with
   ATrpms (and automatically later into rpmrepo.org)? That way we won't
   have further fragmentation and a larger user base to test bits (which
   will be distributed in stable, testing etc repos).

 Of course I'd love to contribute my changes to ATrpms.  Some of the
 small changes I made, such as adding OSLEC to the DAHDI RPMs, might be
 nice for ATrpms users.  I'll whip up some patches against the ATrpms
 sources.

 My problem with ATrpms, though, is that the RPMs make use of many custom
 macros that make them unbuildable outside the ATrpms environment.  I
 understand that might be necessary for RPMs like DAHDI that build kernel
 modules for several versions of several distros, where vanilla specfile
 code would get hairy.  (I think we had this discussion a couple of years
 ago on the ATrpms ML.)  Since I don't have to worry about multiple
 versions of multiple distros in my environment, I prefer to use vanilla
 specfile that will rebuild on anyone's CentOS 5 system.


Alternatively, there's also the RPMS at
http://packages.asterisk.org/centos/which seem to have a nice spread
of options available, including 1.4/1.6
packages, are pretty nicely modularised and seem to be kept pretty fresh...
They do however seem to have some issues that your RPMS (and Axel's) don't
(e.g. why wouldn't an init file be included? and where's the changelog?)...
Perhaps it would be useful to help the digium packager build some better
packages... That would also help with reducing fragmentation, if there were
decent quality 'official' packages available then it would save the time and
effort Axel and the rpmrepo.org folks too as they could in theory base any
extras on those packages rather than needing to maintain the entire set...

d
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Re: [asterisk-users] Asterisk multi-cpu

[D Tucny]

From your figures, it would appear that if you double the load you will be
potentially starting to see problems...

FYI, not sure if it's of use to you... but... The digium tc400b is a
transcoder card that can offload upto 120 channels of transcoding for g729
- ulaw... It's available as PCI only, but, if that's OK, it could be an
alternative to replacing your server... G729 licenses are not needed when
using that card...

There have been posts by some people about having multiple CPU machines but
finding that asterisk's load wasn't spread over those CPUs very well... I'm
not sure if they had something special happening that caused their symptoms,
but, from your dual core machine you should be able to see whether or not
the load is already being spread across the 2 cores OK with your workload...

d

2009/3/27 Mike l...@virtutel.ca

 Thanks that`s great info, and I've already subscribed to the HA mailing
 list.

 I understand call handling takes little CPU, but half my calls are
 transcoded from ulaw to g729 and vice versa.  That seems to take my single
 CPU, dual-core 2.5Ghz machine up to ~35% CPU utilization.  I imagine
 doubling what happens on my server would take me dangerously close to the
 upper limit of good call quality.

 Am I complete off?

 Mike

  -Original Message-
  From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of David Backeberg
  Sent: Thursday, March 26, 2009 18:40
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Asterisk multi-cpu
 
  On Thu, Mar 26, 2009 at 3:06 PM, Mike l...@virtutel.ca wrote:
   Hi,
  
   I know somebody is going to give me the link to the wiki hardware
 pages,
  but
   I can't find the answer there. I'd like to know if, for an Asterisk
 only
   system (nothing else of note running on it), I get a real gain from
  having 2
   CPUs.
  
   Does the amount of traffic/SIP registrations/codec translation possible
   doubles with 2 CPUs? (each quad core E5420 to be precise)? Does it
  increase
   by 50%?  It is only a marginal increase, or none at all?
 
  You don't say anything about your possible kind of usage, so it's
  difficult to provide any specific answer to your question. In general,
  a few things are true:
  * asterisk is multi-threaded
  * linux kernel has nice job schedulers and i/o schedulers
  * if you have more ram, more things will get cached in ram
  * if you have more cpus / cores you can do more things at once as long
  as they aren't all idle waiting for some resource constraint
 
  You need to run a LOT of traffic through a server if it's just
  straight call handling, with a minimum of disk-bound i/o or
  transcoding, before you're going to max out modern hardware. So just
  buy the best server you want to buy, but save some money for a good
  warranty, or buy two servers if that's cheaper than what it would cost
  to be down.
 
  If you want more in-depth discussions on this you probably would
  prefer the asterisk-ha-clustering list:
  http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering
 
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Re: [asterisk-users] incoming call problem from pri

[D Tucny]

2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr

 Hi, i managed to connect to Ericsson MD110 with PRI at last.
 And made a successful call thru asterisk to ericsson.

 But when i try to call from ericsson to asterisk i got an error on
 asterisk side.
 And i couldnt figure out why.

 Here's my extensions.conf about incoming calls.

 [DID_span_1]
 include = DID_span_1_timeinterval_all,${timeinterval_all}
 DID_span_1_timeinterval_all]
 exten = s,1,Goto(default,8000,1)


 I am forwarding all calls coming from PRI trunk to extension 8000 in
 asterisk.
 and what i got on debug of asterisk when i create a call from pbx to
 asterisk..

  Extension '8000' in context 'DID_span_1' from '' does not exist.
 Rejecting call on channel 0/4, span 1


 What can it be the error.??


It would appear you are sending calls to extension 8000, but, as the error
message indicates, there is no extension 8000 in the context you are sending
the call to, there's only an 's' extension...

The s extension isn't a wildcard extension so won't match the inbound call
to 8000... if you had exten = 8000,1,Goto(default,8000,1) in the inbound
context then that would match and the call would be sent to extension 8000
in the default context, which is probably the functionality you are
expecting...

d
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Re: [asterisk-users] Asterisk and PBX internal numbers

[D Tucny]

2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr

 Hi, i know i am asking a lot of questions lately in this forum..sorry
 about that first of all. :)



 Ok, here is the deal..
 I am trying to make a hybrid system with an ericsson MD110 and asterisk.
 Internally we have 4 digit phone extensions on ericsson.. and so in
 asterisk.
 So, what i want to do is to call pbx side without adding 9 or etc to the
 begining of the number from asterisk clients..
 For example if smbody tries to call 1500 first asterisk should look for
 its local extensions and, if it not there it should dial PRI trunk..
 Is there any way for that?


It's pretty simple... e.g.

[context-for-phones]
include = localphones
include = everythingelse

[localphones]
exten = 1500,1,Dial(SIP/1500)
exten = 1501,1,Dial(SIP/1501)
exten = 1502,1,Dial(SIP/1502)

[everythingelse]
exten = ,1,Dial(DAHDI/g1/${EXTEN})

So, your phones would be set up to use context-for-phones, then if a call
was made the localphones context would be checked for a match before the
everythingelse context... If 1500/1501/1502 were dialled, the calls would go
the SIP/1500/1/2, any other 4 digit extension would be sent to the DAHDI
group 1...

d
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Re: [asterisk-users] Asterisk and PBX internal numbers

[D Tucny]

2009/3/20 D Tucny d...@tucny.com

 2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr

 Hi, i know i am asking a lot of questions lately in this forum..sorry
 about that first of all. :)



 Ok, here is the deal..
 I am trying to make a hybrid system with an ericsson MD110 and asterisk.
 Internally we have 4 digit phone extensions on ericsson.. and so in
 asterisk.
 So, what i want to do is to call pbx side without adding 9 or etc to the
 begining of the number from asterisk clients..
 For example if smbody tries to call 1500 first asterisk should look for
 its local extensions and, if it not there it should dial PRI trunk..
 Is there any way for that?


 It's pretty simple... e.g.

 [context-for-phones]
 include = localphones
 include = everythingelse

 [localphones]
 exten = 1500,1,Dial(SIP/1500)
 exten = 1501,1,Dial(SIP/1501)
 exten = 1502,1,Dial(SIP/1502)

 [everythingelse]
 exten = ,1,Dial(DAHDI/g1/${EXTEN})

 So, your phones would be set up to use context-for-phones, then if a call
 was made the localphones context would be checked for a match before the
 everythingelse context... If 1500/1501/1502 were dialled, the calls would go
 the SIP/1500/1/2, any other 4 digit extension would be sent to the DAHDI
 group 1...


Sorry, small correction my example is missing a '_' before  which would
be needed...

d
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Re: [asterisk-users] Asterisk and G.726 Codec

[D Tucny]

2009/3/18 Kevin P. Fleming kpflem...@digium.com

 Le'an Liu wrote:

  My questions:
  1. G.726 16/24/32/40 supported in asterisk-1.6.0.5?

 No. Only G726-32 is supported in all Asterisk versions.


Perhaps the confusion in the voip-info page mentioned is due to the other
G726 rates being supported in files with format_g726?
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Re: [asterisk-users] Good phone near $125

[D Tucny]

2009/3/17 Marc Charbonneau timebandit...@gmail.com

  I was looking at the aastra 9133i, however I was informed that this phone
 is
  no longer supported. What are good phones around the $100 - $125 price
  point? (Need POE)

 I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,
 support PoE and works with 2.5mm headset.
 $110 at voipsupply


While the Polycom IP-330s do work reasonably well, my first impressions with
them was of a cheap light plasticy handset, while that might be a preference
issue, all the other IP phones I've used (Cisco, Linksys and Avaya) had more
substantial handsets that I found more comfortable to use personally and
were preferred by the user base... That said, the Polycom's do the job and
for their price, you get a decent amount of value... I personally probably
wouldn't choose them for myself as the handset is an issue for me, but, in a
reasonable size deployment where cost is a factor, they aren't bad phones...

d
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Re: [asterisk-users] Is it possible to get full callin number from E1?

[D Tucny]

2009/3/12 ssmax ss...@126.com

 Hi all

i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp
when somebody dial  +86 020 87654321 , the asterisk will get the call in
 number by ${EXTEN} variable, but it can only get 87654321, no area code .


As others have mentioned in this thread, this is pretty normal... Telco's
all have different policies on how many digits they will pass for the
dialled number, some may let you specify, but often each telco has an
internal standard... You don't need the rest of the number, but if you're
doing something where it would be useful to have it, you can easily add the
prefix yourself... It's also worth pointing out at this point, the prefix is
either 020 or +86-20 for Guangzhou depending on whether it's written for
national or international use, the 0 is not dialled internationally...



when someone dial  +86 020 87654321 ,  means 4 digits,  the
 phone can call in, and the  ${EXTEN}  is only  87654321 too ,
is it possible to get  full call in number 87654321  in asterisk ?
  thanks


This is confusing... Guangzhou numbers are only 8 digits long + prefixes,
so, +86 20 87654321 would be the number... Where do you get '' from?
What would be more normal would be that you'd by DDIs with your ISDN30
service, such that you'd have for example, a range of 100 numbers say from
87654000 to 87654099, though you can buy more... With this, the number
dialled (or the last 8 digits in your case) would be accessible through the
${EXTEN} variable... I guess it could be possible that your telco is
attempting to offer some other type of service that allows them to issue you
a single number, but, that they will accept as many as four additional
digits that they will pass to you in some way, but, it's impossible to say
what they are doing here and how it's been implemented... I would further
suspect that if this is the case that there would be a very distinct chance
that this would only work with inbound calls from other users of the same
telco, and in China, in the same city...

Within China, telco interoperability can be flaky at best and, from
experience, the fact the telco engineers don't entirely understand their
huawei switches for local problems and for problems over a wider area, the
fact that the national telcos are split into provincial and city
organisations that don't appear to communicate without going through some
circuitous route via the telco's head office both lead to some very long fix
times if there ever is a fix... That said, as long as you have cash, you can
get them to provide almost any service you want, though they won't often
offer anything but the most basic service, you have to ask for it...

d
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Re: [asterisk-users] Outgoing call drops

[D Tucny]

2009/3/12 Danny Nicholas da...@debsinc.com

  Greetings Listers,

   I’m running 1.4.21.2 on SUSE 11.0 with and zaptel
 1.4.12.1 on a TDM400P.  Most of my calls work great, but occasionally we try
 to connect to a customer or vendor external conference call and the call
 will drop after 60-65 seconds unless I have an Answer before the Dial in the
 dialplan.  Isn’t this solution a hack and what would be a better one?
 Thanks in Advance.


Unfortunately, when dealing with analogue lines, hacks are quite often
needed... I'd suggest that in this case something is causing asterisk (or
perhaps the card if polarity reversal is used) to fail to detect the fact
that ringing has ended or an answer signal has been passed when calling
these numbers... As such a timeout occurs...

You'd need to provide more information to pinpoint the exact issue... but...
the only real solution to getting reliable in call event signalling is to
use digital lines (or IP services that use digital lines or all IP) though
some analogue services do a better job using polarity reversals which makes
it much more reliable, the fact that these services are few and far between
makes that not an option for most...

d
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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

[D Tucny]

A 2950 can be configured to limit the speed per port...

I guess the ISP here is operating this way because they are out of the way
and have limited bandwidth themselves, so, they are trying to split up the
bandwidth provided into smaller, more manageable chunks to avoid overloading
things at their end...

In asia here too the ISP that has service in this building has put in 24port
switches, if I ask for ethernet service, I'm told there's no such thing, all
I can order is ADSL, if I order ADSL, I get a 10Mb/s ethernet connection to
the switch, but then internet access is provided over PPPoE limited to 3Mb/s
both ways, I can get additional connections, to the same switches, with
seperate PPPoE accounts, again limited to 3Mb/s... So, at least I'm luckier
than Vikas, but, there is no alternative... There are features I would like,
but, in a monopoly you get what your given...

It's possible to load balance traffic over 4 connections though without any
help from the ISP... It won't be perfectly balanced, but it will do a
reasonably decent job... The options are many though and it depends on what
kit you have... I've done it with cisco routers before without nat where the
ISP was happy to support it and linux firewalls with nat with multiple
ISPs...

d

2009/2/13 Alex Balashov abalas...@evaristesys.com

 Oh--you mentioned in an earlier post that the Cisco switch was installed by
 the ISP, so presumably that is something they consider their CPE as well.

 You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950
 does not have a Layer 3 feature set;  that only comes with MSFCs on
 higher-order Catalysts.  So, they are doing in some fashion other than on
 the switch ports, which is why I asked about the routed interfaces;  does
 anything plugged into a given port have a separate routed interface?

 -- Alex


 On Fri, 13 Feb 2009 04:17:37 -0500, Alex Balashov
 abalas...@evaristesys.com wrote:
 
  This discussion is not making any sense to me.
 
  Just what type of access product is this?
 
  If you have fiber to the premise and are handed Ethernet from there to a
  Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
  Interconnection) type product.  It could also be framed over mid-band
 gear
  over copper at some point in the circuit design and they could be fibbing
  you on the fiber to the premise bit;  the fiber involved may actually
 be
  a remote terminal or mux somewhere in the vicinity.  Either way, if you
  have media converter CPE on your premises, this is an Ethernet product.
 
  If that's so, there's no 512 kbps line.  There is no xDSL.  And there
 is
  no incentive whatsoever to sell copper circuits as Ethernet transport is
  usually more expensive and high-margin product.
 
  Do you have a routed IP interface on your side?  If so, what equipment is
  it on?  It's not the switch, as the switch is Layer 2.
 
 
  On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen
  benny+use...@amorsen.dk benny%2buse...@amorsen.dk
  wrote:
  Vikas topg...@gmail.com writes:
 
  The ISP said that they ran a fiber optic wire to a media box at our
  office and from there there is a RJ45 to the switch. They bring no new
  equipment to our premises each time we provison a new port. Hence this
  upload speed limitation is not due to the copper wire.
 
  So the ISP is being deliberately difficult. I am assuming that their
  motivation is that they want to sell E1's instead of the 512kbps
  lines.
 
  You can fight your ISP by installing various multiplexing equipment,
  but it's an arms race, and they will probably win it -- losing you as
  a customer obviously doesn't worry them, while you're apparently
  willing to go to great lengths to stay with them.
 
  I would recomment just switching to E1 (preferably with a different
  provider). It's that or moving HQ to somewhere sane.
 
 
  /Benny
 
 
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  --
  Alex Balashov
  Evariste Systems
  Web: http://www.evaristesys.com/
  Tel: (+1) (678) 954-0670
  Direct : (+1) (678) 954-0671
  Mobile : (+1) (678) 237-1775
 
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 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Caller ID replacement

[D Tucny]

The hotdesking section of the asterisk book may also be of interest...

d

2009/2/13 David Ruggles da...@safedatausa.com

 Some googling lead me to this:
 http://hans.fugal.net/blog/tag/astdb

 Which looks like it has an answer.

 Thanks all!

 Thanks,

 David Ruggles
 CCNA MCSE (NT) CNA A+
 Network EngineerSafe Data, Inc.
 (910) 285-7200  da...@safedatausa.com



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
 Ruggles
 Sent: Thursday, February 12, 2009 12:24 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Caller ID replacement


 Could you give me an example of how this would look in the dialplan?

 Thanks,

 David Ruggles
 CCNA MCSE (NT) CNA A+
 Network EngineerSafe Data, Inc.
 (910) 285-7200  da...@safedatausa.com



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
 Nicholson
 Sent: Thursday, February 12, 2009 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Caller ID replacement


 On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote:
  I'm working on building a pbx that will allow us to use our cellphones as
  extensions (to some extent)
 
  The dialout is working fine. What I would like to do is have an inbound
  cellphone call appear as if it were an extension. So right now if I call
 in
  from cell #9995551212 the caller id is 9995551212 but if I dial extension
  30013 it will call cell #9995551212. I would like to change the caller id
 so
  9995551212 is changed to 30013 on the inbound call. Doing one is simple
  enough, but I would like have an easy (more or less) way of setting up
 some
  global variables that link the cell phone #'s and extensions and have
 this
  done somewhat automagically.

 I would implement this using the a database (astdb or odbc) containing
 the mapping from cell number to extension.  Then for each call that may
 need callerid modification, you can check the database for the proper
 mapping.  With this method it is also easy to add new mappings.

 --
 Matthew Nicholson
 Digium, Inc. | Software Developer


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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

[D Tucny]

2009/2/13 Vikas topg...@gmail.com

 My questions are:
 1. The black wire coming into the Mc Manstel box is that a fibre optic
 cable ?
 2. What is the Mc Manstel box doing ?
 3. What CISCO router do I need to buy to do bandwidth aggregation at my end
 ?


1) Yes
2) It's stopping you from poking the fibre... the point where they spliced
the fibre coming in to the fibre running to the switch is on that plastic
box...
3) A cisco router with at least 5 ethernet interfaces, which would likely be
expensive, but, look at
http://www.cisco.com/en/US/tech/tk648/tk361/technologies_configuration_example09186a0080950834.shtmlfor
more info...

d
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Re: [asterisk-users] meetme application

[D Tucny]

2009/2/9 邱磊 qiulei...@163.com


 i reload the app_meetme.so in CLI:
  - Reloading module 'app_meetme.so' (MeetMe conference bridge)
   == Parsing 'etc/asterisk/meetme.conf': Found

 All the sip message show that there is no fault, and i dont know why the
 meetme application can't work.

 i have a usable meetme sever and the sip message is:
  Executing MeetMe(SIP/20742-081a8198, 1000|Ap) in new stack
   == Parsing '/etc/asterisk/meetme.conf': Found
   == Parsing '/etc/asterisk/staticmeetme.conf': Found
 -- Created MeetMe conference 1023 for conference '1000'


Your config looks correct, but, one thing that looks a bit odd to me is the
output from your reload of app_meetme, I'm not sure if you typed the output
and made a typo, but, the path to meetme.conf when parsed after the reload
is relative rather than absolute... I'm not sure if this is relevant at all,
but, looks odd... You haven't changed the paths in
/etc/asterisk/asterisk.conf to be relative paths have you?

As I say, this may or may not be relevant... but... I could imagine that if
relative paths from root have been used and asterisk is not launched from
the root then it may have alsorts of issues finding configs... It could be
useful for you to enable debugging output too, not that app_meetme makes
much debugging output, but, it does output a little which may at least
identify the point it's getting to...

d
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Re: [asterisk-users] escaping regular expression

[D Tucny]

2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at

 Hi!

 I am going nuts using REGEXP. I just want to verify if a variable
 contains a valid +E164 phone number.

 These, the the pattern is ^\+[0-9]+

 First I tried:

   Set(pattern=^\+[0-9]+);
   if (${REGEX(${pattern} ${${var}})})

 but that does not work, the backslash is removed, as seen in the log file:

   func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
 http://www.adaanumber.com/)

 So, meanwhile I tried to escape the backslash. I tried:
   Set(pattern=^\\+[0-9]+);
   Set(pattern=^\\\+[0-9]+);
   Set(pattern=^+[0-9]+);

 But always the same result:

   func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)

 How can I solve this problem?


Try something like... pattern=^[+]\{1\}[0-9]+

d
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Re: [asterisk-users] escaping regular expression

[D Tucny]

2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at



 D Tucny schrieb:
  2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
  mailto:klaus.mailingli...@pernau.at
 
  Hi!
 
  I am going nuts using REGEXP. I just want to verify if a variable
  contains a valid +E164 phone number.
 
  These, the the pattern is ^\+[0-9]+
 
  First I tried:
 
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
 
  but that does not work, the backslash is removed, as seen in the log
  file:
 
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
  http://www.adaanumber.com/)
 
  So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
 
  But always the same result:
 
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
 
  How can I solve this problem?
 
 
  Try something like... pattern=^[+]\{1\}[0-9]+

 Are you sure? The \ should be in front of the +


Pretty sure...

exten = *56,1,NoOp(Starting regexp test)
exten = *56,n,Set(pattern=^[+]\{1\}[0-9]+)
exten = *56,n,Set(var=123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))
exten = *56,n,Set(var=+123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))


[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:1] NoOp(SIP/*01-09bd8ff8, Starting regexp test) in
new stack
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:2] Set(SIP/*01-09bd8ff8, pattern=^[+]\{1\}[0-9]+) in
new stack
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:3] Set(SIP/*01-09bd8ff8, var=123456789) in new stack
[Feb  4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(123456789)
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Function result is '0'
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Function result is 'No Match'
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:4] NoOp(SIP/*01-09bd8ff8, No Match)) in new stack
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:5] Set(SIP/*01-09bd8ff8, var=+123456789) in new stack
[Feb  4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(+123456789)
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Function result is '1'
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Function result is 'Match'
[Feb  4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb  4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:6] NoOp(SIP/*01-09bd8ff8, Match)) in new stack

So, the \ is still stripped (ast_app_separate_args removes \), but, it
doesn't matter as the + is bracketed so it's not the first character after
the ^ and so regcomp doesn't fail...

d
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Re: [asterisk-users] Contact lookup

[D Tucny]

2009/2/4 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net


 On Tue, 3 Feb 2009, Geoff Lane wrote:

  Hi All,
 
  Asterisk 1.4.12 on CentOS 5
 
  I'd like to be able to look up each incoming CLI to retrieve an
  associated name, if available, and then pass that to the extensions so
  that they can see both the name and number of the caller. I'm not
  after LDAP or anything else maintained externally, just a contact
  lookup for my system.
 
  I suspect that Astdb could be used for this, as could a relational
  database like MySQL or postgres (accessed via AGI?) Probably simpler
  would be to maintain a text configuration file since I'm only
  concerned about less than a hundred entries initially.
 
  I'd appreciate insight into which is the easiest way to do this, and
  also any pointers to tutorials etc.

 AstDB:

 At it's very simplest:

 exten = s,n,Set(CALLERID(name)=Unknown)
 exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
 exten = s,n,GotoIf($[${name} = ]?endCID)
 exten = s,n,Set(CALLERID(name)=${name})
 exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
 ${CALLERID(all)})

 ... somewhere in the incoming processing. (This is an extract from an
 overly complcated macro I use) Things to check for - a name already being
 present - eg. on an incoming SIP call. No name in the astDB - might want
 to substitute Unknown ..

 All you need to do now is populate the astDB - I use a web interface and
 some php to drive the manager interface...

 My biggest site has just under 300 lookup entries... (Which presents other
 issues with the web interface, but ...)

 I use a slight variant of this...

exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})

Basically the same as yours above (including substitution of Unknown when
not found), but, all on one line...

I've been looking into changing it recently such that where I don't have the
name I can substitute something more useful than Unknown, such as the site,
or for external calls, the country/province/state/city/type/telco/etc,
though that won't be in astdb due to the current 100s of thousands of
rows...

d
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Re: [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i

[D Tucny]

Please get this out of office reply disabled, or at the very least, fixed...
It currently seems to have generated a loop, sending out of office replies
to the out of office replies it's already sent to the asterisk-users mailing
list... It's bad that it sent a reply to the list anyway, but this behaviour
is very very bad...

d

2009/2/4 gust...@utopixnetworks.com

 Me encuentro de vacaciones hasta el proximo 16/02.

 Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com)
 o Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).

 Muchas Gracias,
 Gustavo Scheveloff


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Re: [asterisk-users] Quiet 24 port POE gig switch

[D Tucny]

2009/2/2 Steve Underwood ste...@coppice.org

 Bernd Felsche wrote:
  Ian Cowley i...@moffat.co.uk wrote:
 
 
  Beware PoE switches that can't handle Class 3 (15W) on all ports.
  Most have fans because 24 (or 48) x 15W is hot!
 
 
  That's the power supplied .. which'd be at the far end of the wire.
 
  The efficiency of the PSU plays a big part in the heat dissipation.
  The push to compact dimensions doesn't help ... a 400W or
  thereabouts PSU with 24 independent outputs in 1U height? I suppose
  if the switch were quite deep it could be workable and quiet.
 
  The problem isn't simply of being fanless. But being quiet.
  Preferably below 32 dBA at 1 metres for most offices.
 
  You can do that by using fans other than the tiny, whiney, 40mm fans
  that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
  fans at the back or front, pushing air in (hence the deep
  dimensions), but the top and bottom would need recesses to allow
  sufficient airflow when the positions above and below are filled.
 
 So, size does matter after all. :-)

 24 x 15W = 360W. Its not that big a supply really, and spread across a
 1U case its not that dense a supply. A 360W desktop PC supply can be
 pretty quiet, so its sad none of the 1U chassis supplies are. Probably
 if they used a large impeller fan they could get the noise down. I guess
 they assume these things will be in cupboards or data centres where
 nobody cares. This is a poor assumption.


The problem is squeezing fans in that can push enough air to keep it cool...
For a 1U device, you have only 4.445cm to work with, with a 4mm fan, that
would be 2.2mm of space for casing etc above and below, reasonably tight
already... A quiet 80mm fan as you may find in a PC PSU that puts out
somewhere between 15-20dBA of noise will typically move between 20 and 30
cfm of air... A quiet 120mm fan at the same noise levels would typically
move between 30 and 50 cfm of air and a quiet 40mm at those levels would
move about 5 cfm of air... Obviously, they aren't using quiet 40mm fans...
To get the airflow of the quiet 80mm fans, a 40mm fan has to go very fast
and you're looking at noise levels of approx 40-60dBA, not exactly quiet,
but, that's not all, even if the fan was silent, forcing the air through the
small cramped chassis of a 1U device is going to be noisy...

The assumption made when they make these devices is that the vast majority
of people will put this kit somewhere out of the way in a likely temperature
controlled, reasonably sound insulated environment, with the rest of their
hardware that lives hidden from people... These people will likely prefer
that kit uses the space as efficiently as possible, so, squeezing as much
functionality into as few rack units as possible is important... They have
typically made a good assumption in this I would say... Admittedly, people
who are planning an office for their first time may more commonly neglect IT
hardware, it's requirements (and those of those people around it), from an
IT standpoint, it's a significant pain to deal with, but, in most cases I've
seen, it's something that's considered very carefully if planning an office
in the future...

I suspect the lack of larger quieter units in the market is reflective of
the much lower demand for these, somewhat specialised devices... On the
otherhand, soundproofed rack cabinets that have integrated cooling and look
nice/plain enough that they don't scare people in an office should be
generic enough that there would, I suspect, be sufficient demand from those
that didn't consider IT requirements when fitting out an office to justify
making them...

d
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Re: [asterisk-users] Trunk with Polocom Video Conferencing Unit

[D Tucny]

2009/2/2 Daniel Harper dan...@harper.net.nz

 I was wondering if anyone can help me with a problem we have at one of
 our sites.

 We have setup a Asterisk Trunk to a Avaya PBX, ie ...

   Avaya - Asterisk (1.2.30) - External ISDN Network

 BUT They also have a Polycom VSX 7000 that with some sort of BRI
 converters that plugs into the Avaya.

 The Trunk is working well except for Video Conference Calls. The
 Polocom can receive but not make calls, and the calls that it receives
 drop out every 5 minutes.

 Short of telling them to fork out for a BRI service does anyone have
 any ideas how to rectify the drop-outs? Would 1.4 help? Is there a way
 to plug the Polycom into the Asterisk server directly?

 Any help would be much appreciated.


The Polycom has a number of interface options, quad BRI being one of them...
Your Avaya PBX has some BRI modules with the Polycom probably using 4 of the
ports on one of those modules able to use 8 channels at the same time...

You've not mentioned how the Avaya is connected to Asterisk and you've not
provided any information on messages you are getting on the Asterisk console
when attempting to make outbound calls or when calls drop out on inbound
calls... This would be useful to help you determine what is happening... If
using a PRI type connection between the Avaya and Asterisk, it might also be
useful to get some pri debug output during these tests...

It could, in theory be possible to connect the Polycom directly to Asterisk,
possibly with BRI, possibly through the use of H.323, but the support for
both of those is somewhat less mature than much of the rest of asterisk...

d
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[asterisk-users] Improving asterisk documentation - sources and what the community can do

[D Tucny]

Hi All,

There have been a number of comments recently about a shortage of
documentation on Asterisk, so I wanted to cover briefly the documentation
options available and suggest what they are useful for and how they can be
improved...

Documentation sources:
http://www.asteriskdocs.org/
- Not much there at the moment, due to come back up with new content in
2009...
- Should be a good location for concentrating documentation effort
- links to the Asterisk Book
- asterisk-doc mailing list exists too, but, also seems dead...
- Difficult to see how to contribute at the moment

The Asterisk Book (2nd edition)
- Available to buy and to download for free as a PDF
- Provides a good overview of the concepts for manager/beginner level
- Provides a guide to getting Asterisk up and running
- Refers to a single version of asterisk (1.4)
- Is a static document with a reasonably long release cycle
- Difficult to see how to contribute

Docs that come with asterisk
- README contains basic installation process
- UPGRADE-* covers some of the changes between major versions
- docs/* has a range of docs, some out of date and with holes in it's
coverage
- embedded documentation as seen with core show application and core show
function, good basic reference information, but, not always the most
convenient place to get to it
- Updated documentation can be fed in by the community through patches and
additional docs

www.voip-info.org
- Variable levels of information about different topics, some detailed, some
with hardly any information
- Missing information, no content at all for some
commands/functions/features...
- Inconsistent information, different pages say different things about the
same thing...
- Lack of detail about applicable versions, not always obvious that things
have changed between versions or have been removed or deprecated
- Structure has become a bit messy and could use some attention
- Some large pages that can prove troublesome to update
- Should be able to hold as much information as anyone would want with full
details, examples and highlighted version differences...
- Generally easy for anyone to update if they register an account

So, the easiest way that people could contribute to improving Asterisk
documentation right now would appear to be by improving articles on
www.voip-info.org...

It would still be good to get the documentation process back up and running,
but, until that's there, I'd like to propose that people have a look over
the content at www.voip-info.org and see if there's anything they can see
that could be improved... It would probably be useful to have templates of
some sort for things such as applications and functions, but, I'm not sure
how easy they are to setup with the wiki software in use there, I'll have a
look in a bit, but, if anyone knows, some info on that could be a useful
start...
I've already spent some time trying to update some of the reference info,
such as making a start on updating the applications page, but, there is so
much that could be improved that all together, there will be quite a lot of
effort involved in getting it tidy... I'm going to continue to work on it,
but, if others also dig in it shouldn't take too long to get the reference
documentation there up to a decent level...

If anyone has any other sources of information, or any other suggestions for
improving the documentation, I'd like to see them...

d
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Re: [asterisk-users] Ntework Card

[D Tucny]

This is still very off topic...

Someone's already suggested you look to somewhere for centos help... if you
had, you'd have found this...
http://www.centos.org/modules/newbb/viewtopic.php?forum=39topic_id=10098

Which is an RPM containing an update driver, so you wouldn't have to mess
about poluting your system...

Also, you'd have found documentation which would tell you how to setup a NIC
manually... I'd guess you missed the bit about editing modprobe.conf...

If you ask these question in the right place, you'll probably get much
better help...

d

2009/1/26 David @ULC ucoms2...@gmail.com


 [r...@vicidialnow ~]# cat /etc/sysconfig/network-scripts/ifcfg-eth1
 NAME=
 BOOTPROTO=none
 TYPE=Ethernet
 DEVICE=eth1
 MTU=
 NETMASK=255.255.255.0
 BROADCAST=192.168.0.100
 IPADDR=192.168.0.100
 NETWORK=192.168.0.0
 ONBOOT=no


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Re: [asterisk-users] Auto Detect

[D Tucny]

on CentOS you can use lshal to return which driver is in use...

e.g.

lshal -s |grep pci_ |xargs -n1 lshal -l -u |grep -E
udi|info.product|info.linux.driver

get's a list of items, filters on pci, gets the long output for those pci
devices and outputs lines containing udi, info.product and
info.linux.driver...

results returned like the following (but with more devices
obviously)0...

udi = '/org/freedesktop/Hal/devices/pci_10b7_9200'
  info.udi = '/org/freedesktop/Hal/devices/pci_10b7_9200'  (string)
  info.product = '3c905C-TX/TX-M [Tornado]'  (string)
  info.linux.driver = '3c59x'  (string)
udi = '/org/freedesktop/Hal/devices/pci_10ec_8139'
  info.udi = '/org/freedesktop/Hal/devices/pci_10ec_8139'  (string)
  info.product = 'RTL-8139/8139C/8139C+'  (string)
  info.linux.driver = '8139too'  (string)

d

2009/1/26 bails ba...@westcomuk.com

 Just to reply to myself and top post in the process.

 Its seems this is not the case on CentOS systems my earlier post was the
 part of the output from a box running Debian Lenny.

 lspci -v on a CentOS-5.1 box shows neither Kernel driver in use or
 Kernel modules.

 Bails

 bails wrote:
  Probably not much use to you in this instance, but its always nice to
  know that lspci -v prints the kernel driver in use and kernel modules.
 
  e.g.
 
  10:00.0 Ethernet controller: Broadcom Corporation NetLink BCM5787M
  Gigabit Ethernet PCI Express (rev 02)
Subsystem: Hewlett-Packard Company Device 30c2
Flags: bus master, fast devsel, latency 0, IRQ 220
Memory at d000 (64-bit, non-prefetchable) [size=64K]
Expansion ROM at ignored [disabled]
Capabilities: access denied
Kernel driver in use: tg3
Kernel modules: tg3
 
  Bails
 
  David @ULC wrote:
   lspci
  ~
  00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller
 (rev
  03)
  00:02.0 VGA compatible controller: Intel Corporation 82G35 Express
  Integrated Graphics Controller (rev 03)
  00:02.1 Display controller: Intel Corporation 82G35 Express Integrated
  Graphics Controller (rev 03)
  00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network
  Connection (rev 02)
  00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
  Contoller #4 (rev 02)
  00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
  Controller #5 (rev 02)
  00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI
  Controller #2 (rev 02)
  00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio
  Controller (rev 02)
  00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express
 Port
  1 (rev 02)
  00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express
 Port
  2 (rev 02)
  00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express
 Port
  3 (rev 02)
  00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
  Controller #1 (rev 02)
  00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
  Controller #2 (rev 02)
  00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
  Controller #3 (rev 02)
  00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI
  Controller #1 (rev 02)
  00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2)
  00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface
  Controller (rev 02)
  00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port
 SATA
  IDE Controller (rev 02)
  00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller
 (rev
  02)
  00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port
 SATA
  IDE Controller (rev 02)
  03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller
  04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet
  Adapter (rev 11) (rev 11)
  04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70)
 
 
  but
 
  *service network restart RETURNS me*
 
  *** Shutting down interface eth0: [ OK ]** Shutting down loopback
  interface: [ OK ]** Bringing up loopback interface: [ OK ]**
  Bringing up interface eth0: [ OK ]** Bringing up interface eth1:
  skge device eth1 does not seem to be present,** delaying
  initialization.** [FAILED]*
 
 
 
 
  On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com
 wrote:
 
  Which command to run which will auto detect all hardwares present in
 the
  system ?
  OS : CentOS
  Running Asterisk
 
 
 
  
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 


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 dangerous content by MailScanner, and is
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Re: [asterisk-users] [SOLVED] Nortel IP phone i2002 - DHCP server unreachable

[D Tucny]

Perhaps this would help...

http://blog.michaelfmcnamara.com/2007/10/dhcp-options-voip/

Gives details on the dhcp option string needed for the phones and explains
that without it the phone will not accept a DHCP response...

d

2009/1/24 Joseph syscon...@gmail.com

 Thanks for the input.
 Yes, I have all the correct setting in the phone.  However, it turn out
 that I need to have as DHCP server Nortel BCM (Business Communication
 Manager), whatever
 it is.  It must be some proprietary stuff.

 So the only option for me was to setup IP manually; and it did work
 following this guide:

 http://www.oneconnect.ca/files/userguide.i2002-i2004ConfigurationInstructions.pdf

 --
 #Joseph
 GPG KeyID: ED0E1FB7


 On Fri, 23 Jan 2009, Alexander Lopez wrote:

 1  Can you verify that you have a DHCP server running on that network
 segment?
 2  Can you verify that the Ethernet port on the phone is indeed seeing
 link from the switch?
 3  Have you run wireshark/tcpdump to see if anything is traveling
 to/from the phone?
 
 Alex
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Joseph
  Sent: Friday, January 23, 2009 9:58 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Nortel IP phone i2002 - DHCP server
 unreachable
 
  Is anybody using Nortel IP Phone?
  I have (second hand) Nortel i2002 phone and when it boots I get:
  DHCP server unreachable
 
  F/W version: 0604D9C
 
  My setting:
  DHCP? [0-No, 1-Yes]: 1
  DHCP: 0-Full, 1-Partial: 0
 
  Can any body suggest how to troubleshoot it?

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Re: [asterisk-users] registration problem using asterisk 1.6

[D Tucny]

2009/1/22 Laurent Bonny laurent.bo...@gmail.com

 Hello,

 I am trying to connect an asterisk 1.6 to a trunking plate forme. With
 asterisk 1.4.x I added to sip.conf a line asking for registration in the
 form of:
 register = 
 xx...@domain.com:Password:xx...@domain.comassword%3axx...@domain.com
 @domain.com

 Unfortunately, as you can see,  my usernames have to be of the form
 xxx...@domain.com which means that I had to put 2 @ at the end of my line,
 1 for the username and 1 for the domain.
 In Asterisk 1.6 it doesn't seems to work anymore the @ being a reserved
 sign, and something like this line being impossible.
 Is it a bug from asterisk (as I don't see why I couldn't have a username in
 this form)? Would you know of a way to register my users correctly?


Could you not do something like this?

register = X:passw...@provider

[provider]
type=peer
host=domain.com
fromdomain=domain.com
username=X
fromuser=X
secret=password

Then it would register as xx...@domain.com...

d
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Re: [asterisk-users] OT - Is Netgear ProSafe FS108P with PoE silent ?

[D Tucny]

2009/1/23 Olivier oza-4...@myamail.com

 Hi,

 I need to locate a Asterisk server and a PoE-enabled switch on someone's
 desk.
 I've seen this Netgear ProSafe FS108P with 4 10/100 and 4 10/100 w/PoE
 ports silent enough ?
 Any recommendation ?


Specs on netgear.com say it's silent... 'Acoustic Noise: 0 dB'

d
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Re: [asterisk-users] No Ring on Analog Phone using Rhino ChannelBank in China

[D Tucny]

2009/1/23 Lee, John (Sydney) john@compuware.com


  There's nothing special about analogue phones in China, they are fully

  interchangable with analogue phones elsewhere... Perhaps you have a
  configuration problem, or, hardware problem on the Rhino Channel Bank,

  perhaps the ports are wired the wrong way and the phones care, perhaps
 the  phones have the ringers disabled...

 D, thanks for replying to my problem.


No problem...


 I contacted Rhino and they told me to just reconfigure the T1 line and
 it appears to fix the problem.
 My question is as we made zero changes to the channel bank, why do we
 have to reconfigure it to get it to work?  Do you need to reconfigure it
 every now and then?


I've not used Rhino kit, but, that sounds like a firmware bug that they have
a workaround for...  With any luck it's very infrequent and they'll be
releasing a fix once they've worked out the cause... Sorry I can't help,
might be best to ask Rhino about the details of the problem...

d
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Re: [asterisk-users] Packet8 hacked

[D Tucny]

2009/1/23 Dean Collins d...@cognation.net

  Looks like www.packet8.com has been hacked L

 The phone service is offline as well.

 Anyone else on this list using packet8?


Not using packet8, but, the website looks normal to me...

What are you seeing?

d
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Re: [asterisk-users] No Ring on Analog Phone using Rhino Channel Bank in China

[D Tucny]

2009/1/21 Lee, John (Sydney) john@compuware.com

 I am testing analog phone and fax machine plugged into Rhino Channel
 Bank which is connected to TE412P card.  This site is in China.

 I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4

 I ran into a problem which is analog phone can hear dial tone and can
 make outgoing calls.  Another phone (ether internal or external) can
 call the analog phone ***but the phone does not ring***.  However, if
 the person knows that someone is calling him and picks up the analog
 phone, he will be able to talk to the caller.

 This problem does not happen in other countries which I tested before.

 I have tried distinctive ring tones like [ Dial(Zap/32r5,20) ]  but they
 don't seem to make the analog phone ring.
 I think it has to do with the analog phone doesn't recognize the ring
 voltage generated by Rhino.
 Does anyone have experience with this?
 Do we have to modify the output ring voltage from the channel bank to
 make it work?


There's nothing special about analogue phones in China, they are fully
interchangable with analogue phones elsewhere... Perhaps you have a
configuration problem, or, hardware problem on the Rhino Channel Bank,
perhaps the ports are wired the wrong way and the phones care, perhaps the
phones have the ringers disabled...

d
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

Zeeshan,

I've put them on a server so that you should be able to access them...

http://astinfo.newportcoastsoftware.com/SIPDefault.cnf
http://astinfo.newportcoastsoftware.com/SIPmacaddress.cnf

d

2009/1/21 Zeeshan Zakaria zisha...@gmail.com

 D Tuncy, if you don't mind, can you show me your config files which you are
 using to successfully register phones on two servers. I tried various
 different things, and once it got registered on two servers, but couldn't
 dialout on any. Now it is again back on only one server and I don't remember
 changing any proxy setting. I must be doing something wrong which I don't
 know what it is.

 Zeeshan


 On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote:

 That's not my experience...
 e.g.

 SIP Phone show register

 LINE REGISTRATION TABLE
 Proxy Registration: ENABLED, state: REGISTERED
 line  APR  state  timer   expires proxy:port
   ---  -  --  --
  
 1 111  REGISTERED 115 98  192.168.1.1:5060
 2 111  REGISTERED 115 98  192.168.1.12:5060
 3 ...  NONE   0   0   undefined:0
 4 ...  NONE   0   0   undefined:0
 5 ...  NONE   0   0   undefined:0
 6 ...  NONE   0   0   undefined:0
 1-BU  111  REGISTERED 115 98  192.168.1.1:5060

 Note: APR is Authenticated, Provisioned, Registered

 I can see registers on both servers...

 d

 2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

  From my experience it won't register to the second box, only to the
 first one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me
 yet.

 I have universal setting in SIPDefault.cnf and phone specific settings
 in SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like
 this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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 --
 Zeeshan A Zakaria

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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 Olivier oza-4...@myamail.com



 2009/1/20 Gordon Henderson 
 gordon+aster...@drogon.netgordon%2baster...@drogon.net
 

 On Tue, 20 Jan 2009, Olivier wrote:

  GTalk seems to fill the bill of requirements, though, I don't think it's
  available on Nokia  mobile phones ..

 My Nokia mobile phone has a SIP client built-in which uses Wi-Fi... And
 while it's not perfect, it's actually very usable and works well with
 asterisk.

 Upgrade the phone?


 I agree that native Nokia SIP client is very usable and should be the
 preferred way to call over a WiFi connection.
 I mentioned GTalk as someone mentioned Fring on Nokia.


Unfortunately, I upgraded my phone to one of these new fangled S60 3rd
edition Feature Pack 2 phones (N78 in this case, but N96 has the same
software) where Nokia have removed the SIP client... They've included a SIP
stack and stated that this way third parties can write their own clients...
Options are somewhat limited at the moment...

d
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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 Olivier oza-4...@myamail.com




 One thing to note about fring, the device establishes a connection using
 fring's proprietary protocols to fring servers, fring then establishes SIP
 connections from those servers... So, even if connected to the office Wifi
 connection, you could experience connectivity issues or high latency as a
 result of a potentially long path involved for the traffic to travel...

 d



 So, when in the office, whenever I'm calling someone using SIP and WiFi,
 for both signalling and media, data would travel from mobile phone to WiFi
 access point, then to switch, router, Fring server (at the other end of the
 world), then back to my routeur, and Asterisk server, right ?


Correct...



 If think I'll try to compare this with a XMPP/Jingle-enabled client that
 can be installed in mobile phone (I don't know if such software exist) ...


Not seen anything, though I've mostly been looking for Nokia phones, my boss
has a iPhone though, so something for that wouldn't be bad either...

d
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Re: [asterisk-users] Problem with TDM808

[D Tucny]

If your provider provides any signalling to indicate answer, such as a
polarity reversal, this could be detected easily...

; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
;
;answeronpolarityswitch=yes

This isn't very common though... alternatively, there is the 'HIGHLY
EXPERIMENTAL' call progress detection...

; On trunk interfaces (FXS) it can be useful to attempt to follow the
progress
; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.
;
; Few zones are supported at the time of this writing, but may be selected
; with progzone.
;
; progzone also affects the pattern used for buzydetect (unless
; busypattern is set explicitly). The possible values are:
;   us (default)
;   ca (alias for 'us')
;   cr (Costa Rica)
;   br (Brazil, alias for 'cr')
;   uk
;
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
;
;callprogress=yes
;progzone=uk

Obviously far from ideal, and at least, where I am, unworkable due to the
way that all the telcos have got into providing musical ringing...

The only real solution is to go digital...

d


2009/1/21 Pascal Bruno tipas...@gmail.com

 Is there any way of going around this???  Any tricks, configuration hacks??





 On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote:

 On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
  I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
  1.6.3.  When I try making a call with a .call file, the call goes
  straight to the dialplan and start executing the dialplan even before
  the called party has pick up.  Anybody knows why by any chance?

 That's not a problem with the TDM800 card... it's just a side-effect of
 analog signaling.  For analog calls, the central office doesn't give any
 type of signal when the far end has answered the call, so Asterisk has
 no way of knowing when that happens. For that reason, Asterisk
 immediately treats any outgoing analog call as having been answered.

 --
 Jared Smith
 Digium, Inc. | Training Manager




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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 Olivier oza-4...@myamail.com

 Hi,

 Is anyone using Fring as a SIP client to an Asterisk server ?


Yes, testing it...



 A prospective customer of mine is asking to integrate its iphones with an
 Asterisk server and after googling, I still have some unanswered questions :

 1. Which codecs are available when calling from fring ?


I believe it offers GSM, ilbc, ulaw and alaw...



 2. Is it easy and natural to change your presence status (available, busy,
 ...) with Fring or will users prefer to use another software (bundled with
 iPhones) or to do nothing at all ?


 3. Is it possible to add custom presence status in Fring client ?


On the version running on my nokia phone, there seems to be no obvious way
to change status...



 4. Is it possible and recommended to limit Fring usage to WiFi presence ?


The options on the version I have are: Wifi first, 3G/GPRS first, Wifi only,
3G/GPRS only, Always ask... So, it is possible... It does work over GPRS,
but quality is noticably lower than over Wifi...



 5. Would fring replies to Qualify messages ?


Yes

One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could experience connectivity issues or high latency as a
result of a potentially long path involved for the traffic to travel...

d
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings in
 SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


That all looks fine, though I don't think the ports need quotes...

You do have
proxy_register: 1
too don't you?

If so, check the debug output from sip set debug on, or, sip set debug peer
224 to see if it's reaching the server but not authenticating etc... Also,
you can log into the phone using telnet to check status and restart
registration (and many more things)

d
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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 John Todd jt...@digium.com


 On Jan 19, 2009, at 6:29 PM, D Tucny wrote:

  2009/1/20 Olivier oza-4...@myamail.com
  Hi,
 
  Is anyone using Fring as a SIP client to an Asterisk server ?
 
  Yes, testing it...
 
 
  A prospective customer of mine is asking to integrate its iphones
  with an Asterisk server and after googling, I still have some
  unanswered questions :
 
  1. Which codecs are available when calling from fring ?
 
  I believe it offers GSM, ilbc, ulaw and alaw...
 
 
  2. Is it easy and natural to change your presence status (available,
  busy, ...) with Fring or will users prefer to use another software
  (bundled with iPhones) or to do nothing at all ?
 
  3. Is it possible to add custom presence status in Fring client ?
 
  On the version running on my nokia phone, there seems to be no
  obvious way to change status...
 
 
  4. Is it possible and recommended to limit Fring usage to WiFi
  presence ?
 
  The options on the version I have are: Wifi first, 3G/GPRS first,
  Wifi only, 3G/GPRS only, Always ask... So, it is possible... It does
  work over GPRS, but quality is noticably lower than over Wifi...
 
 
  5. Would fring replies to Qualify messages ?
 
  Yes
 
  One thing to note about fring, the device establishes a connection
  using fring's proprietary protocols to fring servers, fring then
  establishes SIP connections from those servers... So, even if
  connected to the office Wifi connection, you could experience
  connectivity issues or high latency as a result of a potentially
  long path involved for the traffic to travel...
 
  d


 Yes, it's disappointing that Fring doesn't release the media.  I tried
 it a few times, but the latency was unacceptable since the hairpin
 apparently has very long legs from a millisecond-delay perspective.

 I had it working once, but now for some reason I get authentication
 errors (repeated REGISTERs and they don't seem to see my replies) and
 I don't really have the time to work it out.  Anyone having similar
 problems or is this my local problem?


I've just had a look, and can confirm that it is working here... even quite
a while after the fring client has been closed :/

Fring seems to send CR and LF twice in one packet every 10 seconds, not
relevant here, but, a bit odd...

One thing that might be relevant, it does use NAT, at least some of the
time, so asterisk needs to be aware of that... Address combinations I've
seen in the past hour...
91.151.216.12:52931
212.150.129.13:52924 - 172.16.8.15:52924
91.151.216.10:53219
91.151.216.4:52796

d
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Re: [asterisk-users] followme order field

[D Tucny]

2009/1/19 Thomas Stein thomas.st...@knowledgetools.de

 Hello.

 Does someone know what order field means in followme.conf? The Doku says:

 number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
 order in follow-me] ]

 So an example would be:

 number= 123124125,10,?

 It would be nice if someone could enlighten me.


As I understand it, Follow-me can take multiple number lines in the
config... (I've not used follow-me though, this is just from looking at
app_followme.c)

number = 123124125,10,1 ; Would call 123, 124  125 first (all at the
same time as the same syntax in a Dial string would do), trying for 10
seconds
number = 126,10,3 ; Would call 126 third, trying for 10 seconds
number = 127,10,2 ; Would call 127 second, trying for 10 seconds

In the example I've provided, 127 would be called before 126 due to the use
of the order field... If the order was not specified then 126 would be
called before 127 due to the order of the lines in the file...

Again, I've not used it, so not 100% sure that's how it works, but, that's
what it looks like to me... Hope it helps...

d
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

That's not my experience...
e.g.

SIP Phone show register

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line  APR  state  timer   expires proxy:port
  ---  -  --  --
 
1 111  REGISTERED 115 98  192.168.1.1:5060
2 111  REGISTERED 115 98  192.168.1.12:5060
3 ...  NONE   0   0   undefined:0
4 ...  NONE   0   0   undefined:0
5 ...  NONE   0   0   undefined:0
6 ...  NONE   0   0   undefined:0
1-BU  111  REGISTERED 115 98  192.168.1.1:5060

Note: APR is Authenticated, Provisioned, Registered

I can see registers on both servers...

d

2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

 From my experience it won't register to the second box, only to the first
 one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings in
 SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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Re: [asterisk-users] How to hangup a call manually...

[D Tucny]

2009/1/17 Carlos Chavez cur...@telecomabmex.com

I have this call:

 SIP/protel-525512047 default  90445528885371  1 Ringing
 AppDial  (Outgoing Line)   90445528885371  264:24:2
 (None)

I cannot use the soft hangup commando from the CLI because I do not
 know the whole SIP channel string.  What other command can I use to
 terminate this call or to find the complete channel string to put into
 soft hangup?


It looks like you have the entire channel string there, but, if not... The
CLI support tab completion, so

soft hangup SIP/protel-tab

will either fully autocomplete, or, where there are multiple matches, give
you a list of matching channels...

d
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