Re: [asterisk-users] Customized Queuing Strategy

2008-08-04 Thread Benoit Plessis
Alex Balashov a écrit :
 Syed Nasruddin wrote:


   
 1. 10 Call Center Agents.

 2.   All the calls coming in will ALWAYS be routed to specific 5 agents, 
 firstly.

 4. IF ALL the first 5 agents are busy then ONLY then the call will be 
 routed to next 5 Agents.
 

 Set up two queues.  Call Queue() on the first queue - corresponding to 
 #1 - with a rather strict timeout.  Fall back on the second queue.

 More sophisticated strategies require either the modification of the 
 source code for app_queue, or custom queue implementation in AGI
Isn't this possible with agent priority ?




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Re: [asterisk-users] AMI able to call from known endpoint to unknown endpoint?

2008-08-01 Thread benoit plessis
On Thu, Jul 31, 2008 at 05:28:42PM -0700, Stephen Cattaneo wrote:
 Both are sitting behind a Linksys IP PBX (SPA9000).  On the Linksys IP
 PBX I have set the outside number 5000 to connect to 3001.  3002 does
 not have a similar external mapping (this would defeat the purpose of
 the test I am attempting).
 [...] 
 Is it possible (and if yes, how can I do this) to use the AMI's
 originate to call from 5000 to 3001?

Don't you mean to 3002 ?
either that or you will make a loopback call ...

Anyway, if that's what i understood it's impossible. You can see
a PABX a little bit little a network NAT device (well in your specific
problem). Behind the Linksys there is a private network, with one
public adresse, which is a static map to one private adresse.

Well at least it's how your asterisk IPBX will see things.
Basically when using asterisk to connect (call) two extension,
asterisk will dial each extension, and then connect them. Since
he doesn't have any access to your 3002 internal extension, there
is no way this could work.

When calling from 3001 you can reach the 3002 extension, but only 
because you are within the same network, to keep the metaphor

-- 
Benoit

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Re: [asterisk-users] Whitepaper: How and to whom sell VoIP

2008-08-01 Thread Benoit Plessis

As for me i mostly saw spellings mistakes, but that's me :)
Grygoriy Dobrovolskyy a écrit :
 i saw that billing iface somewhere else, maybe i am wrong...

 2008/7/30 Mindaugas Kezys [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

 Hello,

 Based on our own and our clients' experience we compiled short
 manual: How
 and to whom sell VoIP

 Hope it can be useful to some of you also.

 You can download it from our site: http://www.kolmisoft.com




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Re: [asterisk-users] Asterisk Queues problem

2008-08-01 Thread Benoit Plessis
Syed Nasruddin a écrit :

 Hi,

 I have Asterisk 1.4.18 and I have been running call center queues on 
 it. Today it suddenly stopped adding inbound calls to queues. I am 
 facing with following error: _app_queue.c:3939 queue_exec: unable to 
 join queue “myqueue”_

 In extension file:

 Queue(myqueue|t|||120)

 And my agents are joining in following manner:

 Exten = 1001,1,AgentLogin(SIP/1001)

 Exten = 1000,1,AgentLogin(SIP/1000)

 One more thing my asterisk successfully captures the call , it plays 
 music on hold but when it starts to push the call in queue it gives 
 out this error.

 Any one help me out. It’s a production machine.

 Thanks

 Syed nasr

I would recommend upgrading your asterisk to at least 14.20.1
I have had many troubles with queues, SIP and IAX with asterisk 1.4.18 that
have been fixed in the following releases


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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

2008-07-29 Thread benoit plessis
On Tue, Jul 29, 2008 at 01:48:26PM +1000, Lee, John (Sydney) wrote:
 On the box, first of all, I just installed Zaptel 1.4.10.1.
[..]
 
 BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681]
 

Hi, just for (all of) you to know this is a known bug of zaptel  
1.4.11, the firmware upload procedure is taking some time, operating
like a freeze during the process, so this message appears.

But this isn't a real problem, as it doesn't have any consequences
appart from the message.


-- 
benoit

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Re: [asterisk-users] Cisco vs Asterisk

2008-07-25 Thread Benoit Plessis

Al Baker a écrit :

Quote

Yet amazingly (if this is, indeed, a source of amazement for you), CCM 
and other Cisco software can be just as buggy as anything OSS, if not 
worse. 


This is simply NOT TRUE and shows a complete lack of understanding of modern 
software development.
CISCO software is developed in a CMM environment.
It has a formal test methodology and uses Automated Testing on EACH new release 
to ensure that 100% of the software that functioned in the Last Release, 
actually works in this release.
  

So every professional software is now free of bug ??
And every professional team that build thoses software are perfect 
people ???
And every automated process made to validate thoses software are so 
complete that nothing is left out 


Well that doesn't explain the vast quantity of bug fixed in every Cisco 
IOS release, or even the vastly

unfinished CRS-1 product from ... cisco...


Further, there is mandatory soak-testing  for all new software.
Sorry, anyone who wants to compare Professional TELCO GRADE software 
development with Open Source is just Completely and Totally freakin clueless.


Yes, so professional telco grade doesn't have bug fix release ??


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Re: [asterisk-users] Cisco vs Asterisk

2008-07-22 Thread Benoit Plessis
voip crazy a écrit :
 Hello all,

 A client of us, is thinking to migrate their actual PBX to a Cisco
 CallManager. We want to sell him an asterisk box to complement the
 Cisco PBX.
 I think to use asterisk as a Voicemail server (Replazing the Cisco Unity)

 Has asterisk all the functionalities to replace a CIsco Unity server?
 Which functionalities Cisco Unity has than asterisk could cover?
 How could asterisk complement the Cisco Call Manager funcionalities?
   
To answer your questions, one would need to know what exactly are
all the functionalities of a Cisco Unity server,
and more specificaly, what are the needs of your client.

But i'm pretty sure the voip-info wiki can answer the asterisk part...


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Re: [asterisk-users] Cisco vs Asterisk

2008-07-22 Thread Benoit Plessis
[EMAIL PROTECTED] a écrit :
 Call me crazy, but why are you so keen on selling them an Asterisk box
 when you don't even know if its capable of doing what you want to sell
 it for?
   
I won't, i had the same felling ...
 thats kinda scray actually.
   
Yep


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Re: [asterisk-users] GotoIfTime Function

2008-06-27 Thread Benoit Plessis

I would say you have two choices for that:
opt 1, let the carrier provider do the ring
  and then answer, using Wait() or WaitForRing()
opt 2, do it yourself using PlayTones() or Progess()


broadband Voice a écrit :
 Finally did it but only one more problem, I want it to ring once 
 before going to the context or playing the background message.
  

 [day_menu]
 exten = s,1,Answer()
 exten = s,2,Background(welcome-message)
 exten = s,3,Dial(SIP/5960,200,rt)  ; week day goes to 
 Philadelphia Office

 [weekend__menu]
 exten = s,1,Answer()
 exten = s,2,Background(welcome-message)
 exten = s,3,Dial(SIP/5961,200,rt)  ; weekend goes to Delaware Office
  
 [night_menu]
 exten = s,1,Answer()
 exten = s,2,Background(officeclosed)
 exten = s,3,Hangup  ;

 ;incoming
 exten = 1866x,1,GotoIfTime(8:00-18:00|mon-sun|*|*?day_menu,s,1)
 exten = 1866x,n,Goto(night_menu,s,1)





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Re: [asterisk-users] iax2 trunk becomes unreachable (asterisk 1.4.21)

2008-06-22 Thread benoit plessis
On Sat, Jun 21, 2008 at 05:30:51AM -0700, Vieri wrote:
 Hi,
 
 I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk.
 iax2 show peers on both boxes seem to show that all's fine (Status OK on 
 qualify=yes peer).
 voip1 is an Asterisk 1.2.27 production server.
 voip2 is an Asterisk 1.4.21 experimental server in the same gigabit LAN.

Have you tried another 1.4.xx version ?
There is a few problems introduced by this release and i already experienced 
this exact problem.

-- 
Benoit

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[asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan

2008-06-17 Thread Benoit Plessis

Is it possible on a TE220p to deactivate the hardware echo canceler at 
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could 
give better results don't you think ?

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Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan

2008-06-17 Thread benoit plessis
On Tue, Jun 17, 2008 at 10:56:06AM -0500, Kevin P. Fleming wrote:
 Benoit Plessis wrote:
  Is it possible on a TE220p to deactivate the hardware echo canceler at 
  will ? (With a function in the dialpan for example)
  example for fax SDA ,beeing able to shutdown the echo canceler could 
  give better results don't you think ?
 
 All echo cancelers using Zaptel/DAHDI already disable themselves when
 FAX or modem communications are used, based on reception and detection
 of the CED tone that FAX machines and modems generate to make that
 happen. You can tell this happened by looking at the channel in Asterisk
 using 'zap show channel' or 'dahdi show channel' as it will show you
 that the echo canceler was disabled automatically.
 

Karamba !

It mean than that my last option is to put all three digium cards in one box 
and hope,
that it'll fit and that it'll work better :(

(actually i'm using one server with two cards (TE220, B410) which is linked 
with an (slinear)
IAX peer to another server with one TDM800 to talk to the fax machine.

-- 
Benoit


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[asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres

2008-06-16 Thread Benoit Plessis


Hi,

I'm having trouble with a TE220p PRI card and (outbond) caller 
identification.

Previously with usecallingpres=no everything was Ok, one small 
difference between the
BRI (B410p) was that the callerid needed to be stripped from one number.

But then came the need to make hidden calls, and so to enable 
usecallingpres and
SetCallerPres().

if running SetCallerPres(prohib) then the end call get 'private number' 
which is what we want
but with  SetCallerPres(allowed) or SetCallerPres(allowed_not_screened) 
i'm not able to get the
number i want, i got the line identification number in any case (even 
with a callerid of 10 digits)

any idea ?

my zapata.conf:
8
[trunkgroups]
[channels]
context=from-rnis-t2-dys
language=fr
switchtype=euroisdn
signalling=pri_cpe

callwaiting=no
threewaycalling=no
callprogress=no
busydetect=no

pridialplan=unknown
prilocaldialplan=dynamic
priindication=outofband

internationalprefix=00
nationalprefix=33
localprefix=
privateprefix=
unknownprefix=
relaxdtmf=yes

hidecallerid=no
usecallingpres=yes

echocancel=yes

faxdetect=incoming
immediate=no
group=1
channel = 1-15
8

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Re: [asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres

2008-06-16 Thread Benoit Plessis

Benoit Plessis a écrit :

Hi,

I'm having trouble with a TE220p PRI card and (outbond) caller 
identification.


Previously with usecallingpres=no everything was Ok, one small 
difference between the

BRI (B410p) was that the callerid needed to be stripped from one number.

But then came the need to make hidden calls, and so to enable 
usecallingpres and

SetCallerPres().
  

...

Well, it was solved by changing

prilocaldialplan=dynamic
  

to prilocaldialplan=unknown


Is there some good technical documentation about PRI lines somewhere ?
there is a lot of voodoo magic in configuring thoses lines with asterisk ...

--
Benoit

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[asterisk-users] handling SIP trunk with limited concurent calls

2008-06-05 Thread benoit plessis
Hi,

Now that we have a working asterisk server, i'm looking
toward cost optimization :)

We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.

My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.

I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.

So, is there a way with the DIALSTATUS variable to detect a 'full' peer 
?

-- 
Benoit


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Re: [asterisk-users] handling SIP trunk with limited concurent calls

2008-06-05 Thread Benoit Plessis
Gordon Henderson a écrit :
 On Thu, 5 Jun 2008, benoit plessis wrote:

   
 Hi,

 Now that we have a working asterisk server, i'm looking
 toward cost optimization :)

 We are actually testing a SIP provider, which has an interessting
 limitation: each account support at max only two concurrent calls.

 My problem is how to combine multiple accounts and fail back to PSTN
 lines if all accounts are 'full'. I've added a call-limit=2 in the
 sip.conf entry, but i dont really now how to use it in the dialplan.
 ChanIsAvail() was my first try but didn't work.

 I've tried chaining Dial() calls:
  Dial(SIP/line1/${EXTEN})
  Dial(SIP/line2/${EXTEN})
  ...
 but when an error condition occurs (busy/unavailable/whatever) it
 dial the same number on every line, which can take a while at the end.

 So, is there a way with the DIALSTATUS variable to detect a 'full' peer
 ?
 

 Yes.

 You need to check for CONGESTION.

 something like:

n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup

n(tryNext),Dial(SIP/line2/${EXTEN})

 But do check that the SIP provider does indeed return CONGESTION ... (You 
 may not need the call-limit=2, if they check for you, then if at a later 
 date, they increase the limit, then you don't need to change anything)

 Gordon
   
Isn't there a risk of getting a CONGESTION message from the other party ?

benoit


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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

2008-06-05 Thread Benoit Plessis

Brent Davidson a écrit :

...I wonder why more vendors haven't adopted IAX yet?

Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX
as SIP is more mature and reliable in asterisk and zoiper,

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Re: [asterisk-users] handling SIP trunk with limited concurent calls

2008-06-05 Thread Benoit Plessis

Benoit Plessis a écrit :

Gordon Henderson a écrit :
  

On Thu, 5 Jun 2008, benoit plessis wrote:

  


Hi,

Now that we have a working asterisk server, i'm looking
toward cost optimization :)

We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.

My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.

I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.

So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?

  

Yes.

You need to check for CONGESTION.

something like:

   n,Dial(SIP/line1/{EXTEN})
   n,Noop(Dial line1 failed - we got ${DIALSTATUS})
   n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
   n,Hangup

   n(tryNext),Dial(SIP/line2/${EXTEN})

But do check that the SIP provider does indeed return CONGESTION ... (You 
may not need the call-limit=2, if they check for you, then if at a later 
date, they increase the limit, then you don't need to change anything)


Gordon
  


Isn't there a risk of getting a CONGESTION message from the other party ?

benoit

  
Another problem i foresee is long delay in dialing sequence when 
asterisk will have to dial using 4/5 account

before having a working channel

i think i should look after another sip provider

--
Benoit

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Re: [asterisk-users] Asterisk on iPhone

2008-05-19 Thread Benoit Plessis

You might be looking for that instead  http://sip.free.fr/index.html.en

Andrea Cristofanini a écrit :
 Hi
 I just saw this now !
 does the microphone and speaker works ?
 Can you use it like softphone for recive calls ?
 Regards  Andrea
 C F ha scritto:
   


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Re: [asterisk-users] Not hearing first prompts

2008-05-16 Thread Benoit Plessis
Alan Lord wrote:
 Unfortunately not. It is the same if we use our Siemens DECT/SIP 
 handsets or the Ekiga softphone...

 I recall having this problem once before and that it went away when I 
 changed from Ekiga to Twinkle. When I get chance, I will re-install 
 Twinkle and see if that exhibits the same problem.
 
I've always had this kind of problems with ekiga, the first 2/3 seconds 
of sound are
of bad quality (cutted, metallic, ..) and always missing one or two 
seconds, and this last
for at least 3 years. this remind me that i still hasn't verified if 
this as been reported as a bug
to ekiga team

it is the reason why i'm using twinkle as a phone client, and twinkle 
isn't a good client
(weird translation, usability  inexistant) but at least audio is 
good and it can ring on
a sound card and phone using another one.

Anyway instead of doing a Wait(), i used a Anwser() + 
Playback(silence/1) to get around this kind
of thing. Since it's most probably problem while decoding first bunch of 
audio sample, using Wait()
won't help.



-- 
Benoit Plessis  +33 6 77 42 78 32
[EMAIL PROTECTED] +33 4 67 28 06 96


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Re: [asterisk-users] queue problem

2008-05-13 Thread benoit plessis
On Wed, May 14, 2008 at 01:27:38AM +0800, Rilawich Ango wrote:
 I have a queue with the following setting.
 total queue member =30, autofill=1, timeout=25, monitor_format=wav49
 asterisk 1.4.18
 In busy hour, the loading of CPU reaches over 300%.  At that moment,
 all members are occupied and many calls are waiting in the queue.
 There will be choppy and line cut at such high CPU loading.

Hi,
I was having huge problems with AsteriskNow 1.0.1 which is packaged
with asterisk 1.4.18(.1? not sure). Most of them came with the 
deploiment of our support center call queue. With only 2/3 agents and
max 6/10 simultaneous calls the system goes wazaa and eat the 4 cores
of the xeon 1.6 cpu, users gets stucks in 'in use' state (while normaly
this flags doesn't work) and everything goes from bad to worse.

I've rebuild it using etch/amd64 and manual build of asterisk 1.4.20rc2 
(1.4.19.2 wasn't out and i have some IAX calls) and now everything is
fine.


 My questions:
 1. What is the max capacity of a server to handle a queue in term of
 queue member and calls?
Do you use IAX2 ? there is major improvement in this with 1.4.19.2 / 
1.4.20.

 2. After every 25s, the call will be switched from agent to another
 agent.  Can I do something, say execute a CLI or shell command before
 it switches to another agent?

-- 
Benoit

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Re: [asterisk-users] Problem with SIP Subscription Status

2008-05-12 Thread Benoit Plessis
Hach Segal a écrit :
 Hello All,

 I've been having some intermittent trouble with an Asterisk 1.2.10 
   
Before anything else did you tried an updated asterisk 1.2
The last one is 1.2.28 or something like that, and there has been
a lot of security patches, and fixes since your version.

Did you look through the changelog / bugs tracker to see if your
problem has already been reported ?

-- 
Benoit Plessis


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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-10 Thread Benoit Plessis
Patrick a écrit :
 Can you please tell me where you found the backport?

 Thanks,
 Patrick
   
Russel already sent the links:
 Yes, this is also from 1.6, but an unsupported backport of DEVICE_STATE(),
 exists, as well.

 http://www.asterisk.org/node/48325

 http://www.asterisk.org/node/48360
   


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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-10 Thread Benoit Plessis
Steve Totaro a écrit :

 Depending on the phone, the simplest and easiest way to handle this is
 to set the phone to not accept call waiting.

   
That's not always possible, if you use IAX2 then that's not even an 
option (at least on the PBX side):
http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent
Or if you want your queue agent to be joinable with internal calls while 
in communication with a queue
member.


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[asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-09 Thread Benoit Plessis

Hi,

I'm using a Queue in asterisk with IAX2 peers and my agents are also 
doing outbond calls.
Actually we are using a GROUP() function like that to prevent users from 
beeing Dialed while
in communication:
exten = s,1,gotoif($[${GROUP_COUNT([EMAIL PROTECTED])}=0]?:busy)
exten = s,n,Set([EMAIL PROTECTED])
...

But this system fill the asterisk console with queue dial attempt, and 
even with our small system
it make debugging a pain in the a**.

So i'm wondering if someone already as made a dialplan function that 
could toggle the 'Use' flag of
an agent ? or if this kind of function would be integrated into the core 
if i build it ?

Regards,
benoit

CCed to asterisk-dev but i'm not subscribed so please make me in copy.


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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-09 Thread Benoit Plessis
Russell Bryant a écrit :
 Benoit Plessis wrote:
   
 So i'm wondering if someone already as made a dialplan function that 
 could toggle the 'Use' flag of
 an agent ? or if this kind of function would be integrated into the core 
 if i build it ?
 

 This is a slightly different approach, but have you seen the state interface
 code that is in Asterisk 1.6?  There is a backport of the code for 1.4 
 floating
 around somewhere, I think.  It allows you to specify a different device for a
 queue member that app_queue will use to determine the state of an agent.  So,
 you can still list a Local channel for dialing, but Asterisk will look at the
 state of SIP/myphone, for example, to know whether the agent is busy or not.
   
ok but since we are using IAX2 with ZoIPer for sendurl() handling this 
won't help
 Alternatively, if you would like to control the usability of an agent through
 the dialplan, then you could use the DEVICE_STATE() function to create a 
 custom
 device state.  Then, you could list your custom device as what app_queue
 should look at before attempting to call the agent
This is more interesting :)
Is it from 1.6 too ?



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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-09 Thread Benoit Plessis
Benoit Plessis a écrit :
 Russell Bryant a écrit :

 Alternatively, if you would like to control the usability of an agent 
 through
 the dialplan, then you could use the DEVICE_STATE() function to 
 create a custom
 device state.  Then, you could list your custom device as what 
 app_queue
 should look at before attempting to call the agent
 This is more interesting :)
 Is it from 1.6 too ?

don't bother, i found the backport



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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-09 Thread benoit plessis
On Fri, May 09, 2008 at 10:49:02AM -0400, Watkins, Bradley wrote:
 Russell Bryant wrote:
  
 
 One problem with that cunning plan is that using custom device states
 doesn't work.  The code for handling device state changes in app_queue
 is looking for a forward-slash in the device name, and returns if it
 doesn't find one:

Why not modifying func_devstate instead to support Custom/ ?

btw why is it limited to some channel name ? it would be nice 
to use the Agent/xxx channel directly

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Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

2008-05-09 Thread Benoit Plessis
Russell Bryant a écrit :
 I don't see why this wouldn't help.  You just list the IAX2 peer as the device
 Asterisk uses to determine the state of the agent.
   
Well i've read elsewhere that only SIP peers did support the use flag ?


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Re: [asterisk-users] Asterisk in Production ?

2008-05-07 Thread Benoit Plessis
Tzafrir Cohen a écrit :
 On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote:
   
 Tzafrir Cohen a écrit :
 
 On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

   
   
 Here it is, but since the AsteriskNow release has stripped the binary
 i fear it won't be of much use:
 
 
 Is there any -debug package for asterisknow's asterisk package?

 On RedHat they are generated automatically. On Debian they require some
 extra settings, and has been present in recent Asterisk packages (the
 asterisk-dbg package) but not in all of the smaller modules packages.

   
   
 Nope, already tried this before posting
 but nothing like that appears on conary
 

 I looked again at http://rbuilder.rpath.com/ and searched for the
 package asterisk.

 It does seem to have a subpackage called asterisk:debuginfo.
   
I'm not able to install it but i'll look further, conary is a tricky 
software to say the least
   
 anyway, i'll be migrating on a debian asap, since i now this
 much better and the advantages of AsteriskNow keep reducing
 

 Off topic:
 That is not to say you should not try Debian ASAP ;-) 
   
Well i tried a debian/lenny with an mISDN patched for 2.6.24
but it lead to kernel panic / server reboot after 4/5 calls on the B410p.
No problem on the T220b but i need both cards ...

I think i'll have to reinstall an debian/etch and either try the 
packaged asterisk 1.2
or manually build an 1.4 + zaptel + misdn.
Everything i was looking away from when i initially choosed asteriskNow

 To help you with that, here's a live CD:
 http://updates.xorcom.com/iso/

   



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Re: [asterisk-users] mISDN on Debian Lenny

2008-05-07 Thread Benoit Plessis
Philipp Kempgen a écrit :
 Benoit Plessis wrote:

   
 Well i tried a debian/lenny with an mISDN patched for 2.6.24
 

 Are those patches available somewhere? Pointers?

 Regards,
   Philipp Kempgen

   

It's a patch i got from the gentoo portage site, should be made of some 
mISDN commit
in the git tree. but I don't recommend using them, i got two kernel 
panic and a hard reboot after 4/5 calls
http://kambing.ui.edu/gentoo-portage/net-dialup/misdn/files/misdn-2.6.24.diff


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Re: [asterisk-users] Realtime status feature - user feedback needed

2008-05-07 Thread Benoit Plessis
Tilghman Lesher a écrit :
 On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote:
   
 So all together - I'm saying there could be really simple interface
 for all this - no troubles with locking of lists or keeping persistent
 connections. Why would user application need to take care of all this,
 if DB engine can do that.
 

 Your question leads to this question:  why don't you create a proxy
 application that listens on AMI and populates a database outside of Asterisk,
 then do all your queries to that database?  That would provide exactly the
 same functionality, but it would not require a single change to the Asterisk
 codebase.  You could even contribute that application back as something
 in the contrib/scripts subdirectory.

   
I second that,
If there is already a way to do things, why adding another one,
especialy if it's for caching reasons.
While we cannot say that asterisk fall into the KISS rule, it's not
a reason to let it grow.

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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
lordfuknowsyou a écrit :
 Vinícius Fontes wrote:
   
 I use 1.4.18 with no problems. We have quite a few users(125 total 
 between branches), but the call volume at the most has been around 15 
 active calls at a time.
   
Any IAX2 phone or mostly SIP ?
Do you use Call Queues ?

We have less user than that, less concurrent call but quite a few 
crash/deadlocks


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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
Steve Totaro a écrit :
 On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
   
  Any IAX2 phone or mostly SIP ?
  Do you use Call Queues ?

  We have less user than that, less concurrent call but quite a few
  crash/deadlock

 Try SIP only if you can and report back.  I think you will confirm
 what is pretty much a silent consensus (even among Digium Devs).
   
Hi, that's what i was planning seeing all thoses answers.
We initialy choosed IAX2 for the sendurl() support but
i'll set-up a test periode in SIP-only to compare.

 Thanks,
 Steve Totaro
   
Thanks to you


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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
Tilghman Lesher a écrit :
 On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
   
 lordfuknowsyou a écrit :
 
 Vinícius Fontes wrote:

 I use 1.4.18 with no problems. We have quite a few users(125 total
 between branches), but the call volume at the most has been around 15
 active calls at a time.
   
 Any IAX2 phone or mostly SIP ?
 Do you use Call Queues ?

 We have less user than that, less concurrent call but quite a few
 crash/deadlocks
 

 Have you reported these issues on the bugtracker?

   
Well, the problem is finding usefull data to report.

I've 4 core dumps thats show differents things:

two seems to be related to ControlPlayback:
#0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
#1  0x0809c579 in ast_readframe ()
#2  0x0809defc in ast_streamfile ()
#3  0x0805e786 in ast_control_streamfile ()
#4  0xb698be5c in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#5  0x08298700 in ?? ()
#6  0xb470aec0 in ?? ()
#7  0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#8  0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
#9  0x in ?? ()


One is pretty generic:
#0  0x0809c9bc in ast_closestream ()
#1  0x08085d91 in ast_hangup ()
#2  0x080cd3d8 in pbx_builtin_setvar_helper ()
#3  0x080cf08e in ast_pbx_outgoing_exten ()
#4  0x080fde65 in ast_inet_ntoa ()
#5  0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0
#6  0xb703667e in clone () from /lib/tls/libc.so.6


and the latest is thread/iax2 related:
#0  0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
#1  0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#2  0x0079 in ?? ()
#3  0x in ?? ()
#4  0xb547a148 in ?? ()
#5  0x080f0508 in ast_sched_add_variable ()
#6  0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#7  0x0012 in ?? ()



But my main problem is when the system just froze,
it start mostly by the Queue not working anymore, with member stuck in 
'in use' stack (should not happen
with IAX2 agent IIRC, given that we had to build macros using GROUP() to 
detect in use IAX2 agent)
Then the console (asterisk -rcTvvv) start to freeze (completion doesn't 
work, message stop from being displayed
and even command output is lost).

And i'm reading http://www.asterisk.org/developers/bug-guidelines which 
speak of using SVN trunk version of asterisk,
thing i'm not really eager to try on a live system...




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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
Tilghman Lesher a écrit :
 On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
   
 Tilghman Lesher a écrit :
 
 On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
   
 lordfuknowsyou a écrit :
 
 Vinícius Fontes wrote:

 I use 1.4.18 with no problems. We have quite a few users(125 total
 between branches), but the call volume at the most has been around 15
 active calls at a time.
   
 Any IAX2 phone or mostly SIP ?
 Do you use Call Queues ?

 We have less user than that, less concurrent call but quite a few
 crash/deadlocks
 
 Have you reported these issues on the bugtracker?
   
 Well, the problem is finding usefull data to report.

 I've 4 core dumps thats show differents things:

 two seems to be related to ControlPlayback:
 #0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
 #1  0x0809c579 in ast_readframe ()
 #2  0x0809defc in ast_streamfile ()
 #3  0x0805e786 in ast_control_streamfile ()
 #4  0xb698be5c in ?? () from
 /usr/lib/asterisk/modules/app_controlplayback.so
 #5  0x08298700 in ?? ()
 #6  0xb470aec0 in ?? ()
 #7  0xb698c1fc in ?? () from
 /usr/lib/asterisk/modules/app_controlplayback.so
 #8  0xb698c1fa in ?? () from
 /usr/lib/asterisk/modules/app_controlplayback.so
 #9  0x in ?? ()
 
 

 I'd love to see a 'bt full' on this one.
   
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:

#0  0xb705b658 in strcasecmp () from /lib/tls/libc.so.6
No symbol table info available.
#1  0x0809c579 in ast_readframe ()
No symbol table info available.
#2  0x0809defc in ast_streamfile ()
No symbol table info available.
#3  0x0805e786 in ast_control_streamfile ()
No symbol table info available.
#4  0xb698be5c in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#5  0x08298700 in ?? ()
No symbol table info available.
#6  0xb470aec0 in ?? ()
No symbol table info available.
#7  0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#8  0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#9  0x in ?? ()
No symbol table info available.
#10 0x in ?? ()
No symbol table info available.
#11 0x in ?? ()
No symbol table info available.
#12 0x0bb8 in ?? ()
No symbol table info available.
#13 0x2f727669 in ?? ()
No symbol table info available.
#14 0x65696c63 in ?? ()
No symbol table info available.
#15 0x2f73746e in ?? ()
No symbol table info available.
#16 0x6a6e6f62 in ?? ()
No symbol table info available.
#17 0x2d72756f in ?? ()
No symbol table info available.
#18 0x6e656962 in ?? ()
No symbol table info available.
#19 0x756e6576 in ?? ()
No symbol table info available.
#20 0x6568632d in ?? ()
No symbol table info available.
#21 0x6f702d7a in ?? ()
No symbol table info available.
#22 0x62726577 in ?? ()
No symbol table info available.
#23 0x6974756f in ?? ()
No symbol table info available.
#24 0x2d657571 in ?? ()
No symbol table info available.
#25 0x76726573 in ?? ()
No symbol table info available.
#26 0x73656369 in ?? ()
No symbol table info available.
#27 0x696c632d in ?? ()
No symbol table info available.
#28 0x00746e65 in ?? ()
No symbol table info available.
#29 0x0001 in ?? ()
No symbol table info available.
#30 0xb470af20 in ?? ()
No symbol table info available.
#31 0x081aa084 in ?? ()
No symbol table info available.
#32 0x001b in ?? ()
No symbol table info available.
#33 0x0025 in ?? ()
No symbol table info available.
#34 0x0028 in ?? ()
No symbol table info available.
#35 0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#36 0x in ?? ()
No symbol table info available.
#37 0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#38 0x0829c4a8 in ?? ()
No symbol table info available.
#39 0x0bb8 in ?? ()
No symbol table info available.
#40 0x in ?? ()
No symbol table info available.
#41 0xb470aec0 in ?? ()
No symbol table info available.
#42 0x in ?? ()
No symbol table info available.
#43 0xb698c1fc in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#44 0xb698c1fa in ?? () from 
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#45 0x in ?? ()
No symbol table info available.
#46 0x in ?? ()
No symbol table info available.
#47 0x in ?? ()
No symbol table info available.
#48 0x in ?? ()
No symbol table info available.
#49 0x08298700 in ?? ()
No symbol table info available.
#50 0xb705b631 in strcasecmp () from /lib/tls/libc.so.6
No symbol table info available.
#51 0x080c8740 in pbx_substitute_variables_helper ()
No symbol table info available.
#52 0x080cd170 in pbx_builtin_setvar_helper ()
No symbol table info available.
#53 0x080cf08e in ast_pbx_outgoing_exten ()
No symbol

Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
Steve Totaro a écrit :
 On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
   
 lordfuknowsyou a écrit :

 
 Vinícius Fontes wrote:
   
  
   I use 1.4.18 with no problems. We have quite a few users(125 total
   between branches), but the call volume at the most has been around 15
   active calls at a time.
  
  Any IAX2 phone or mostly SIP ?
  Do you use Call Queues ?

  We have less user than that, less concurrent call but quite a few
  crash/deadlocks

 

 Try SIP only if you can and report back.  I think you will confirm
 what is pretty much a silent consensus (even among Digium Devs).

 Thanks,
 Steve Totaro

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I've tried SIP only but i already got one 'stuck' Queue member:
   Members:
  Local/[EMAIL PROTECTED] with penalty 10 (dynamic) (In use) has taken 1 
calls (last was 45 secs ago)
  Local/[EMAIL PROTECTED] with penalty 20 (dynamic) (Not in use) has taken 
no calls yet
   Callers:
  1. Zap/10-1 (wait: 0:18, prio: 0)

[May  6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one 
is answering queue 'support' (1/0/0)
asterix*CLI core show channels
Channel  Location State   
Application(Data)
SIP/rtournier-081ef2 (None)   Up  Bridged 
Call(Local/[EMAIL PROTECTED]

but the other end of the bridged call is long gone



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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Benoit Plessis
Tzafrir Cohen a écrit :
 On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

   
 Here it is, but since the AsteriskNow release has stripped the binary
 i fear it won't be of much use:
 

 Is there any -debug package for asterisknow's asterisk package?

 On RedHat they are generated automatically. On Debian they require some
 extra settings, and has been present in recent Asterisk packages (the
 asterisk-dbg package) but not in all of the smaller modules packages.

   
Nope, already tried this before posting
but nothing like that appears on conary

anyway, i'll be migrating on a debian asap, since i now this
much better and the advantages of AsteriskNow keep reducing

as a matter of fact i already now that some thing that doesn't work 
under AstNow
(my siemens sip hardphones, and my SIP provider (Keyyo) at least) work 
with the
debian packaged asterisk.
Well for the sip provider it's not that it doesn't work, more than the 
only way to have some
sound is to use the 'm' flag of the Dial() command to have the moh 
played during the ringing.
Given that, i got some sound when the call is established ...

-- 
Benoit



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[asterisk-users] Passing Dial URL argument through Asterisk

2008-05-05 Thread Benoit Plessis

Hi,

I'm able to send an HTML Frame to an IAX2 phone like Zoiper, Using 
Dial(IAX2/...,,,http://site/uri;)
but now i need to do this

Inbound Call == Zap == Asterisk-1 [main diaplan, build the needed 
Url ] == IAX2 == Asterisk/2 == IAX2 == Phone

By using a tcp trace i can see that the HTML frame is send to Asterisk/2 
but there is no ${URL} variable set
and so i don't know how to transfer it to the queue and then to the phone.

Any idea ?
Did asterisk even parse the inbound HTML frame ? i can't find evidence 
of this in the source code (rapid glance).

Regards,

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Re: [asterisk-users] Strange CLI behaviour

2008-05-02 Thread Benoit Plessis
lokotes2 a écrit :
 Hi,
 I'm using Asterisk 1.4.17 and 1.4.19 versions, some time ago I've
 noticed that cli command 'core show channels' does not show all data.
 It returns only header or one line of data.
 After that, auto completition of commands (hitting TAB) freezes cli...
 Does anybody has the same problem?
 regards,
 Lokotes.


   

Do you use any zaptel cards ?
We have two cards in our Asterisk server (B410p (misdn) and TE220p 
(zapata))
and this kind of problem seem to happen since the day we started to use 
the TE220p
(auto completion freeze and then the console start to fail big time, no 
more logging message
and then even restart command doesn't work).

regards


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