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Re: Parking calls

It is not a parking solution.

Sebastian wrote:
> Any idea? Please I need advice.
> 
> Thanks!
> 
>  
> 
> From: asterisk-users-bounces <at> lists.digium.com
> [mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Sebastian
> Sent: lunes, 01 de diciembre de 2008 11:58 p.m.
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Parking calls
> 
>  
> 
>  
> 
> Hi,
> 
>  
> 
> How can I park a call from dialplan and get going??
> 
>  
> 
> Example:
> 
>  
(Continue reading)

SIP | 2 Dec 21:52

Re: OT: What do you guys think of this?

Doug wrote:
> At 04:03 12/2/2008, Benny Amorsen wrote:
>  >Doug <Doug <at> NaTel.net> writes:
>  >
>  >> "Net Neutrality" is great in principle.  But ISP's need to
>  >> somehow control those few percentage of users who suck down
>  >> a huge majority of the bandwidth.  It's dollars and cents.
>  >
>  >Yes, just like the airlines need to somehow control those users who
>  >keep showing up to the flight they booked, every single time! It's
>  >impossible to do overbooking with customers like that, so we need to
>  >find ways of punishing them.
>
> What happens if everyone who owns a car drives
> it at the same time?  Owns a telephone and
> uses it at the same time?
>
>
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>   

If everyone who owns a car drives it at the same time, there's lots of 
traffic. You know who gets blamed? The right people -- the people to 
create the infrastructure. Drivers aren't blamed for driving their cars 
when they want to as long as they do it legally as prescribed by the 
(Continue reading)

Tobias Wolf | 2 Dec 19:20

Re: Problem with Bridge Application

Right after sending the email, the solution came to me. I have fooled 
myself: A ManagerEventListener kicked in an issued an HangUp Action on 
the second channel right after the Bridge ...

The Bridging workes perfectly after fixing the EventListener.

Have a nice day, i will go home and hit myself a little bit ...

Tobias Wolf schrieb:
> Hi,
> 
> i am running Asterisk 1.6.0-beta4 and i have some trouble with the 
> Bridge-Application.
> 
> Here is what i want to do:
> 1) Caller A calls an extension and is connected to an AGI-Script.
> 2) Doing stuff and originating a second call per Manager Interface
> 3) Call will be set to an extension with MusicOnHold
> 4) Caller A hears MusicOnHold
> 5) Meanwhile, the second call is established and is also connected to an 
> AGI Script
> 6) Doing stuff
> 7) Since we have transported the channel name of the first call to the 
> agi script we can execute the Bridge Application in order to bridge the 
> two channels.
> 
> After executing the Bridge-Application MusicOnHold is stopped on the 
> first call, but the second call in HungUp immediatly.
> 
> On the Asterisk CLI i see that the correct channel names are issued.
(Continue reading)

Dave Fullerton | 2 Dec 19:22

Re: Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released

Is anyone else having difficulty compiling 1.6.0.2?

It bombs out when compiling manager.c

manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732: error: (Each undeclared identifier is reported only once
manager.c:1732: error: for each function it appears in.)
make[1]: *** [manager.o] Error 1
make: *** [main] Error 2

I see a reference in the 1.6 changelog that refers to SENTINEL not 
existing in 1.6.0

2008-06-27 01:09 +0000 [r125648-125684]  Mark Michelson 
<mmichelson <at> digium.com>

  * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined
    in 1.6.0

-Dave

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(Continue reading)

Danny Nicholas | 2 Dec 17:59

Re: Paging, Polycom and whispers

You can send an IM to the phone with a text message.  Assuming that the
phone has more than 1 line and at least one is open, the call should go
through without effecting the existing call.  To do this from the dialplan,
you could set up something like this:

Exten => 411,1,Dial(SIP/100,1)
Exten => Sendtext(You have a call on park 701)
Exten -> hangup(}

This also assumes that the polycom has presence enabled.

-----Original Message-----
From: asterisk-users-bounces <at> lists.digium.com
[mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 02, 2008 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging, Polycom and whispers

Mike wrote:
> Hi,
> 
>  
> 
> Is there a way to page a Polycom phone that is already in use (if, of
> course, the call isn't on speakerphone already)?
> 

I've never been able to find a way. Any attempt I made either put the 
existing call on hold to auto-answer the page or the page just rang at 
the phone and then caused other issues.
(Continue reading)

Daniel Hazelbaker | 2 Dec 17:19

Re: CDR Design

On Dec 2, 2008, at 7:01 AM, Grey Man wrote:

>> On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy <murf <at> digium.com> wrote:
>> Everyone--
>>
>> I've just made some major changes to the CDRfix2.rfc.txt
>> file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
>> to accommodate the Leg approach instead of a
>> channel-based approach.
>>
>
> Hi murf,
>
> I've got a couple of points (as always) from the new design.
>
> First one would be the generation of CDRs when putting a call on hold.
> I don't think that should occur. When a call is put on hold Asterisk
> never changes the endpoints of a call all it does is possibly change
> the media to one or both of the call ends. CDRs are about call
> endpoints not about media transitions. In SIP terms putting a call on
> hold is no different to changing codecs both operations are re-INVITES
> and are irrelevant as far as CDRs and billing go.

While I agree with your reasoning, I really like the idea of the CDR  
showing HOLD states.  It allows me to generate a report on how often  
people are on hold.  If I see that the incoming calls to my  
receptionist spend 15% of the time on hold, that means something to  
me.  If someone doesn't care to know the hold states, they (or their  
script) can just ignore the HOLD CDR records.  I don't see that it  
would impact any final numbers to just skip them, you still get the  
(Continue reading)

Tilghman Lesher | 2 Dec 16:29

Re: func_odbc and hash problem

On Tuesday 02 December 2008 01:21:46 Giedrius Augys wrote:
> Hello,
>
>   Now I'm testing func_odbc and hash. My configurations are:
>
> func_odbc.conf
> [GETNUMBER]
> dsn=sqlserver
> ;mode=multirow
> ;rowlimit=10
> readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
> WHERE number=${SQL_ESC(${ARG1})}
>
> extensions.conf
> exten => s,1,Ringing
> exten => s,n,Wait(4)
> exten => s,n,Answer
> exten => s,n,Set(NUMERIS=37037210602)
> exten => s,n,Set(HASH(RESULTATAS)=${ODBC_GETNUMBER(${NUMERIS})})
> exten => s,n,Verbose(1, Number is  ${HASH(RESULTATAS, number)}.)
> exten => s,n,Verbose(1, Realus 1  ${HASH(RESULTATAS, real_number1)}.)
> exten => s,n,Verbose(1, Realus 2  ${HASH(RESULTATAS, real_number2)}.)
> exten => s,n,Verbose(1, Statusas  ${HASH(RESULTATAS, status)}.)

Kill the space after the comma.  You're looking for fields whose names are
" number" and " status", which, of course, don't exist.

--

-- 
Tilghman

(Continue reading)

Olivier | 2 Dec 16:15
Favicon

1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?

Hi,

1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ?
I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies.

2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read :
"Also, try using:

 t38_udptl=yes
 t38pt_rtp=no
 t38pt_tcp=no

... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. "

But above, you can read
"[general]
t38pt_udptl = yes "

Has this parameter name changed between 1.4 to 1.6 from t38_udptl to t38pt_udptl ?
A asterisk remains silent when I add an unknown parameter "foo=bar", it would perfect if someone could point the right name (t38_udptl or t38pt_udptl).

Regards

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Peter Galiovsky | 2 Dec 16:06

1.4.22 crashing on Solaris in ast_dynamic_str_thread_build_va

Hello,

Asterisk 1.4.22 keeps crashing on Solaris 5.10 i386.
ast_dynamic_str_thread_build_va() seems to be passed some kind of
garbage (see attached dbx output) which ultimately brings down the
whole process. As a workaround, I've set the debug level to 0 for now.
Should I submit this as a bug?

Thanks for any help. Best,
Peter
t <at> 91 (l <at> 91) terminated by signal SEGV (no mapping at the fault address)
0xfed1587c: strlen+0x000c:      movl     (%eax),%edx
Current function is ast_dynamic_str_thread_build_va
 1354           res = vsnprintf((*buf)->str + offset, (*buf)->len - offset, fmt, ap);
(dbx) where
current thread: t <at> 91
  [1] strlen(0x0), at 0xfed1587c
  [2] _ndoprnt(0xfe8eb5aa, 0xfc5188e4, 0xfc518130, 0x0), at 0xfed6db66
  [3] vsnprintf(0x81fdbcc, 0xb8, 0xfe8eb55c, 0xfc5188e4, 0x81542b0, 0xfedbf000), at 0xfed70c9b
=>[4] ast_dynamic_str_thread_build_va(buf = 0xfc518178, max_len = 1024U, ts = 0x814a9a0, append = 0,
fmt = 0xfe8eb55c "Feature interpret: chan=%s, peer=%s, code=%s, sense=%d, features=%d
dynamic=%s\n", ap = 0xfc5188e4 "çÓ$^H^?Ë$^HÀ\x8aQü^A"), line 1354 in "utils.c"
  [5] ast_log(level = 0, file = 0xfe8ea4cd "res_features.c", line = 1147, function = 0xfe8ea2ab
"ast_feature_interpret", fmt = 0xfe8eb55c "Feature interpret: chan=%s, peer=%s, code=%s, sense=%d,
features=%d dynamic=%s\n", ...), line 807 in "logger.c"
  [6] ast_feature_interpret(chan = 0x827bc10, peer = 0x826a3b0, config = 0xfc518d50, code = 0xfc518ac0
"1", sense = 1), line 1147 in "res_features.c"
  [7] ast_bridge_call(chan = 0x827bc10, peer = 0x826a3b0, config = 0xfc518d50), line 1626 in "res_features.c"
  [8] dial_exec_full(chan = 0x827bc10, data = 0xfc51bbe0, peerflags = 0xfc519af4, continue_exec =
(nil)), line 1780 in "app_dial.c"
  [9] dial_exec(chan = (nil), data = (nil)), line 1834 in "app_dial.c"
  [10] pbx_extension_helper(c = (nil), con = 0xfc51de18, context = 0x827bd90 "outbound_nextra", exten =
0x827bde0 "421912345678", priority = 7, label = (nil), callerid = 0x81751f8 "421212345678", action =
E_SPAWN), line 35 in "strings.h"
  [11] __ast_pbx_run(c = (nil)), line 2317 in "pbx.c"
  [12] pbx_thread(data = (nil)), line 2621 in "pbx.c"
  [13] dummy_start(data = (nil)), line 912 in "utils.c"
  [14] _thr_setup(0xfec6ba00), at 0xfed944c7
  [15] _lwp_start(0x0, 0xb8, 0xfc5181bc, 0xfedbf000, 0xfc518114, 0x0), at 0xfed947b0
(dbx) threads
      t <at> 1  a  l <at> 1   ?()   LWP suspended in  __pollsys()
      t <at> 3  a  l <at> 3   dummy_start()   LWP suspended in  __pollsys()
      t <at> 4  a  l <at> 4   dummy_start()   sleep on 0x8150a20  in  __lwp_park()
      t <at> 5  a  l <at> 5   dummy_start()   LWP suspended in  __pollsys()
      t <at> 6  a  l <at> 6   dummy_start()   sleep on 0x818d7cc  in  __lwp_park()
      t <at> 7  a  l <at> 7   dummy_start()   sleep on 0x818e90c  in  __lwp_park()
      t <at> 8  a  l <at> 8   dummy_start()   sleep on 0x818fa4c  in  __lwp_park()
      t <at> 9  a  l <at> 9   dummy_start()   sleep on 0x8190b8c  in  __lwp_park()
     t <at> 10  a l <at> 10   dummy_start()   sleep on 0x8191ccc  in  __lwp_park()
     t <at> 11  a l <at> 11   dummy_start()   sleep on 0x8192e0c  in  __lwp_park()
     t <at> 12  a l <at> 12   dummy_start()   sleep on 0x81da874  in  __lwp_park()
     t <at> 13  a l <at> 13   dummy_start()   sleep on 0x81db95c  in  __lwp_park()
     t <at> 14  a l <at> 14   dummy_start()   sleep on 0x81dca44  in  __lwp_park()
     t <at> 15  a l <at> 15   dummy_start()   sleep on 0x81ddb2c  in  __lwp_park()
     t <at> 16  a l <at> 16   dummy_start()   LWP suspended in  __lwp_park()
     t <at> 17  a l <at> 17   dummy_start()   LWP suspended in  __lwp_unpark()
     t <at> 18  a l <at> 18   dummy_start()   LWP suspended in  __pollsys()
     t <at> 19  a l <at> 19   dummy_start()   LWP suspended in  ___nanosleep()
     t <at> 20  a l <at> 20   dummy_start()   sleep on 0xfd925420  in  __lwp_park()
     t <at> 22  a l <at> 22   dummy_start()   sleep on 0x8150840  in  __lwp_park()
     t <at> 87  a l <at> 87   dummy_start()   LWP suspended in  __pollsys()
     t <at> 89  a l <at> 89   dummy_start()   LWP suspended in  __pollsys()
o>   t <at> 91  a l <at> 91   dummy_start()   signal SIGSEGV in  strlen()
(dbx) thread -info t <at> 91
        Thread t <at> 91 (0xfec6ba00) at priority 0
        state: active on   l <at> 91
        base function: 0x80f03b4: dummy_start() stack: 0xfc51e000[245760]
        flags: DETACHED|SUSPENDED
        masked signals: HUP INT PIPE TERM WINCH
        Currently active in strlen
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Re: OT: What do you guys think of this?

On December 1, 2008 07:21:33 pm Doug wrote:
> Hmmm.  When our users are pounding the network
> with BitTorrent traffic, we just shut them down
> and wait for them to complain.  It's against our
> Acceptable Use Policy, and causes all sorts of
> VOIP headaches.

As someone who is the technical lead for several ISPs, it is my professional 
opinion that you haven't a clue how to run such a thing.

Torrent does not interfere with VOIP on a well-designed network any more than 
FTP or web browsing.

Honestly, hire a competent admin to set up and run your infrastructure.  If 
torrent's killing VOIP, that means that adding more VOIP will also kill it.  
Or "excessive" web browsing.

Thank God I'm not one of your customers.

-A.

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Kevin P. Fleming | 2 Dec 13:41

Re: Is HPEC compliant with B410P ?

Olivier wrote:

> As latest asterisk-libpri-dahdi is introducing dahdi support of B410P,
> can we use High Performance Echo Canceling addon with B410P ?*

Yes, DAHDI echo cancellers work with any DAHDI supported interface.

--

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Gmane