[Asterisk-Dev] Emprego

Mon Aug 22 07:03:49 MST 2005

Estamos precisando contratar uma pessoa que conheça bem Linux e se já tiver
conhecimentos de ATK será melhor ainda.

Salario a combinar, de preferencia enviem uma pretensao salarial.

Pra trabalhar em SP - Capital.

OBS. Tem que ter firma aberta pois será contrato.

-----Mensagem original-----
De: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] Em nome de
asterisk-dev-request at lists.digium.com
Enviada em: segunda-feira, 22 de agosto de 2005 10:23
Para: asterisk-dev at lists.digium.com
Assunto: Asterisk-Dev Digest, Vol 13, Issue 64

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Today's Topics:

   1. libpri-s makefile (Tzafrir Cohen)
   2. Re: (no subject) (Tzafrir Cohen)
   3. Re: SIP channels not cleared (Olle E. Johansson)
   4. Re: SIP channels not cleared (John Todd)
   5. Re: Help IP phone project (mirza sahib)
   6. Re: Help IP phone project (alex at pilosoft.com)
   7. Re: Help IP phone project (Matt Riddell)
   8. REGEX Function (Alessio Focardi)
   9. Re: Right place to plug in a CSTA(partial)	implementation
      (Apollon Koutlides)
  10. Re: Help IP phone project (Steve Underwood)

----------------------------------------------------------------------

Message: 1
Date: Sun, 21 Aug 2005 21:57:20 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: [Asterisk-Dev] libpri-s makefile
To: Asterisk Developers list <asterisk-dev at lists.digium.com>
Message-ID: <20050821185719.GV31737 at xorcom.com>
Content-Type: text/plain; charset=us-ascii

Hi

I'm looking at libpri's makefile. There aeem to be a number of potential
problems that probably won't bother me (and thus I can't file a patch, as I
can't test it) but seem wrong:

The makefile seems to have been patched by a Solaris guy. It seems to have
the assumption that you're either using Solaring and installing to
/usr/local or using Linux and nor really using ldconfig at install time
(-n: just symlink, don't update cache. Handy for package builders) and
installing files under /usr .

Fine with me for my package. But what if you just want to install it to
/usr/local ? Shouldn't that be the default?

Anyway: I also saw there: 

  ifeq ($(PROC),sparc64)
  PROC=ultrasparc

Does this actually work? Last time I tried something like that, make
complained about redefining values and ignored the second assignment. There
is actually no need to change PROC there, as the new value is only used once
in the makefile: in the next line (still inside that same ifeq).

-- 
Tzafrir Cohen     icq#16849755  +972-50-7952406
tzafrir.cohen at xorcom.com  http://www.xorcom.com

------------------------------

Message: 2
Date: Sun, 21 Aug 2005 22:27:31 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [Asterisk-Dev] (no subject)
To: asterisk-dev at lists.digium.com
Message-ID: <20050821192731.GW31737 at xorcom.com>
Content-Type: text/plain; charset=us-ascii

On Sat, Aug 13, 2005 at 12:54:34PM +0800, Steve Underwood wrote:
> What's up with this mailing list?  Since the problem last week where 
> mail
> stopped for several days, any posting to this list works, but also 
> generates this message back to me:
> 
> This is the Postfix program at host lists.digium.com.
> 
> I'm sorry to have to inform you that your message could not be 
> delivered to one or more recipients. It's attached below.

[with the error below a message from Digium's list server about a mail
forwarding loop]

And they keep on coming...

-- 
Tzafrir Cohen     icq#16849755  +972-50-7952406
tzafrir.cohen at xorcom.com  http://www.xorcom.com

------------------------------

Message: 3
Date: Sun, 21 Aug 2005 21:46:07 +0200
From: "Olle E. Johansson" <oej at edvina.net>
Subject: Re: [Asterisk-Dev] SIP channels not cleared
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <4308D9FF.4010501 at edvina.net>
Content-Type: text/plain; charset=ISO-8859-1

Chee Foong wrote:
> Hello,
> 
> I have tested with the vendor, they send a CANCEL instead of BYE. 
> Another problem came out:
> 
> When the other end send CANCEL, asterisk response with "487 Request 
> Terminated" then a "200 OK" for the CANCEL. The problem now is the 
> other end expect "200 OK" first then "487 Request Terminated". This 
> make asterisk retransmit the "487 Request Terminated" because it did 
> not get an ACK. The second "487 Request Terminated" that asterisk sent 
> got an ACK back from the other end. Call terminated, but again sip 
> channels stays not cleared.
> 
> I have been scratching my head for the past 2 weeks, trying to figure 
> out who is actually sending the wrong stuff.
> 
Please turn on SIP history and try to capture the history of one of those
hanging channels.

/O

------------------------------

Message: 4
Date: Sun, 21 Aug 2005 13:48:10 -0700
From: John Todd <jtodd at loligo.com>
Subject: Re: [Asterisk-Dev] SIP channels not cleared
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <p060204d8bf2e85071bf3 at loligo.com>
Content-Type: text/plain; charset="us-ascii" ; format="flowed"

At 11:33 AM +0200 on 8/21/05, Olle E. Johansson wrote:
>John Todd wrote:
>>  At 7:06 AM +0200 on 8/19/05, Olle E. Johansson wrote:
>>
>>>  Chee Foong wrote:
>>>
>>>>   OK, I under stand.
>>>>   So, can this be considered a bug in asterisk?
>>>>   Since it knows how to response to a BYE, it should also know it's  
>>>> time to
>>>>   clear the channel.
>>>
>>>
>>>  The real fault here is that the other end issues a BYE when we have 
>>> no  session set up by  INVITE/200 OK/ACK - to cancel a pending 
>>> INVITE you use CANCEL, not BYE.  That is a bug, please ask your 
>>> vendor to look up CANCEL in the SIP rfc.
>>>
>>>  And yes, we should be able to handle faulty devices better, but 
>>> will  concentrate our energy on being able to improve the way we 
>>> handle  devices that actually support basic SIP according to the 
>>> standard. ;-)
>>>
>>>  /Olle
>>
>>
>>
>>  This problem could perhaps could be resolved by implementation of  
>> session-timers on the Asterisk side, assuming that the UAC also  
>> supported (or at least did not crash on) such timers.
>>
>>  http://www.faqs.org/rfcs/rfc4028.html
>>
>>  If Asterisk sent re-INVITEs after the Session-Expires: duration, 
>> then it
>>  (Asterisk) could close channels which did not respond.  I would think
>>  that this would be something that could be set on a per-peer basis or
>>  globally.
>>
>>  I believe my previous tests with Asterisk showed that Asterisk 
>> supported
>>  Session-Expires: in a non-harmful way (i.e.: did not crash) but Asterisk
>>  did not seem to have any "hooks" for generating a Session-Expires:
>>  header or creation of timers.  Does anyone have any alternate
>>  information?  It's been a year or so since I experimented with equipment
>>  using session-timers.
>>
>...and you haven't seen my bug report with a patch for SIP timers 
>either? Not session timers, but as a starting point an implementation 
>of the standard T1 timer for retransmits. When that is done, SIP timers 
>would not be a bad thing to add.
>
>/O

Actually, I've been running your patch for about a week, and I've 
just re-patched with CVS-HEAD.  (One chunk fails, but it's an easy 
fix.)

Session-timers would be an excellent thing to add, as they will 
eliminate many of the "phantom channel" problems people have.  At the 
same time, it should be noted that not all equipment handles 
session-timers correctly, so an optional per-peer flag should be 
available as well as the global value.

Perhaps something like:

session-timer=[<seconds>,[yes/no/always]]
min-session-timer=<seconds>
max-session-timer=<seconds>

If session-timer=yes, then accept whatever the other end puts as the 
session-timer as long as it meets min-session-timer and 
max-session-timer.  If "no", then don't accept a session-timer.  (Is 
this worth having?)   If session-timer=always then the session should 
be refused unless a session-timer agreement can be made.

Sound like a reasonable set of values?  Then, it's just a simple 
matter of giving coffee to a programmer and having them do it!  ;-)

JT

------------------------------

Message: 5
Date: Mon, 22 Aug 2005 10:12:05 +0500 (PKT)
From: mirza sahib <wasim at convergence.com.pk>
Subject: Re: [Asterisk-Dev] Help IP phone project
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <Pine.LNX.4.63.0508221006450.31920 at bali.cbs.com.pk>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

On Sat, 20 Aug 2005, Prakash N wrote:

> Iam Prakash doing my project in Design and development of Low cost
> IP-Phone. Iam planning to use single channel IP phone with minimal 
> features.

As some one who went down a similar road a year (or is it two now) back 
and learnt a hell of a lot, but didn't get to the final line very well 
(crawling, panting, with bloody knees wasn't the objective) ... I'd also 
recommend you try the PA168 chipset which has its source available ...

Asterisk, the PBX, on a teeny RISC won't really help you with a phone, 
since * beauty is in its multi-functionality, switching, codec/protocol 
translation, call handling and interfacing to other apps on a *NIX box .... 
none of which you really require on a phone ... what you need is a stable, 
tested and reliable IAX or SIP stack, like others have mentioned on this 
list ... good luck

--
wasim h. baig | principal consultant | convergence pk | +92(300)8508070
GnuPG  fingerprint = F3B5 2B00 F502 51FE 53E3  9607 5257 77B8 AA0D 4AD0

------------------------------

Message: 6
Date: Mon, 22 Aug 2005 01:10:20 -0400 (EDT)
From: alex at pilosoft.com
Subject: Re: [Asterisk-Dev] Help IP phone project
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID:
	<Pine.LNX.4.44.0508220109320.30191-100000 at bawx.pilosoft.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII

On Mon, 22 Aug 2005, mirza sahib wrote:

> As some one who went down a similar road a year (or is it two now) 
> back and learnt a hell of a lot, but didn't get to the final line very 
> well (crawling, panting, with bloody knees wasn't the objective) ... 
> I'd also recommend you try the PA168 chipset which has its source 
> available ...
Very true.

Too bad that there is no manufacturer interested in producing a non-ghetto 
looking PA168 phone. All I need is some blinkenlighten and softkeys, and 
rest can be managed with software...

------------------------------

Message: 7
Date: Mon, 22 Aug 2005 19:10:37 +1200
From: Matt Riddell <matt.riddell at sineapps.com>
Subject: Re: [Asterisk-Dev] Help IP phone project
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <43097A6D.4090605 at sineapps.com>
Content-Type: text/plain; charset=ISO-8859-1

alex at pilosoft.com wrote:
> Very true.
> 
> Too bad that there is no manufacturer interested in producing a 
> non-ghetto
> looking PA168 phone. All I need is some blinkenlighten and softkeys, and 
> rest can be managed with software...

How many units you looking at?

It could be cost effective to make something custom if enough people were
interested.

Any votes for an Open Source development of a phone with Open Source
firmware?

-- 
Cheers,

Matt Riddell
_______________________________________________

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

------------------------------

Message: 8
Date: Mon, 22 Aug 2005 10:00:13 +0200
From: Alessio Focardi <afoc at interconnessioni.it>
Subject: [Asterisk-Dev] REGEX Function
To: asterisk-dev at lists.digium.com
Message-ID: <166449162.20050822100013 at interconnessioni.it>
Content-Type: text/plain; charset=us-ascii

Hi,

anyone can write down a working example of a regex fuction ?

I'm using this syntax

Gotoif($[${REGEX("/B/" | "A")}=1]?20)

But function always return 1, even if I write

Gotoif($[${REGEX()}=1]?20)

Tnx for any help !

-- 
Best regards,
 Alessio                          mailto:afoc at interconnessioni.it

------------------------------

Message: 9
Date: Mon, 22 Aug 2005 13:12:04 +0300
From: Apollon Koutlides <apollon at planewalk.net>
Subject: Re: [Asterisk-Dev] Right place to plug in a CSTA(partial)
	implementation
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Cc: dhetzel at m1global.com
Message-ID: <4309A4F4.3070507 at planewalk.net>
Content-Type: text/plain; charset="iso-8859-7"

dorn hetzel wrote:

>I need a CSTA interface for Asterisk.  At present, I don't need *all* 
>of CSTA, just the parts that will be used by an existing application. I 
>don't see any completed or nearly so implementations out there (Am I 
>missing one or more?).
>
>My present thoughts are to do this as a "translating proxy" which will 
>listen to and respond to CSTA messages and then communicate with
>Asterisk using the manager protocol.   Anybody willing to (a) confirm
>that this seems reasonable?, (b) throw rotten fruit and tell me how I 
>should be doing it?, (c) offer to help?
>  
>
I would guess that the right place to do such a job would be within the 
astman-proxy project... that would also give you call-control facilities 
over multiple asterisk boxen, as well as relieve you from having to 
write the proxy code. We here ("here" is  a Greek telco) are currently 
considering implementing a Call Control interface in that manner, we 
just haven't decided yet which one this should be (TCAP and CSTA are 
among these) - there might even be some hope of effort-saving if project 
timings converge.
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Message: 10
Date: Mon, 22 Aug 2005 21:23:38 +0800
From: Steve Underwood <steveu at coppice.org>
Subject: Re: [Asterisk-Dev] Help IP phone project
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <4309D1DA.80102 at coppice.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

mirza sahib wrote:

> On Sat, 20 Aug 2005, Prakash N wrote:
>
>> Iam Prakash doing my project in Design and development of Low cost
>> IP-Phone. Iam planning to use single channel IP phone with minimal 
>> features.
>
>
> As some one who went down a similar road a year (or is it two now)
> back and learnt a hell of a lot, but didn't get to the final line very 
> well (crawling, panting, with bloody knees wasn't the objective) ... 
> I'd also recommend you try the PA168 chipset which has its source 
> available ...

People keeping saying the PA168 code is open, but it does not appear to 
be true. The only source I have seen available is a small fragment of 
the code needed to make a phone. The rest is supplied as compiled 
libraries. I'd love to hear it has been fully opened up, but so far I 
have seen no evidence of that.

Regards,
Steve

------------------------------

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End of Asterisk-Dev Digest, Vol 13, Issue 64
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