Re: [asterisk-dev] libpri: reverse charging

2007-11-11 Thread Rod Dorman
On Sunday, November 11, 2007, 09:13:54, Vinícius Fontes wrote:
> Sorry, I didn't express myself right. I would like to allow some  
> COLLECT calls based on their caller id (for example employees on  
> field, but not customers). The caller id part is easy, nothing that a  
> GotoIf won't handle... the collect call part is the difficult -- or  
> impossible -- here.

Here  in  the  States when a collect call comes in they announce who the
call is from and you're asked if you want to accept it.

Is it different in Brazil?

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Re: [asterisk-dev] Application timeouts, Periodic and Scheduled Announcements

2007-08-29 Thread Rod Dorman
On Wednesday, August 29, 2007, 15:00:40, Russell Bryant wrote:
>...
> ; Cancel announcements in case the call ended early
> exten => s,n,NoOp(Cancel beep after ${STOP_ANNOUNCE(${ANNOUNCE_ID1})} seconds)
> exten => s,n,NoOp(Cancel timeup after ${STOP_ANNOUNCE(${ANNOUNCE_ID2})} 
> seconds)
> ; Turn off application timeout
> exten => s,n,Set(TIMEOUT(application)=0)

What would be the behaviour if the STOP_ANNOUNCE's and the TIMEOUT
weren't done?

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Re: [asterisk-dev] The New CDR system

2007-04-02 Thread Rod Dorman
On Monday, April 2, 2007, 06:09:41, Tim Panton wrote:
> On 1 Apr 2007, at 18:07, Rod Dorman wrote:
>> On Sunday, April 1, 2007, 04:15:16, Tim Panton wrote:
>>> On 31 Mar 2007, at 17:17, Rod Dorman wrote:
>>>> On Saturday, March 31, 2007, 09:43:53, critch wrote:
>>>>> On Sat, 2007-03-31 at 10:59 +0300, Tzafrir Cohen wrote:
>>>>>>...
>>>>>> What you do is re-implement TCP (or TCP/with a bit of SSL).  
>>>>>> While it might work, ,it is not as good as the original.
>>>>  ...
>>>> I can understand Tzafrir's point of view. The IAX2 protocol draft
>>>> states
>>>>  "Full frames are sent reliably, so all full frames require an
>>>>  immediate acknowledgment upon receipt"
>>>> whereas  TCP  utilizes a sliding window protocol that doesn't
>>>> require an ACK for every individual transmission.
>>>
>>> Actually if you cared to it would be legitmate for an IAX
>>> implementation to ACK a number of fullframes by just acking the
>>> _last_ one, this implicitly acks all the earlier full-frames in that
>>> call.
>>
>> I'm  having  difficulty  finding  where  in  the  IAX2  draft that is
>> specified.  Its  not  stated in "6.11.1. ACK acknowledgement Message"
>> where I would expect to find it.
>>
>> The closest thing I can find is in "7. Message Transport" where it  
>> says:
>>" ... And if the message is a reliable message, the incoming
>> message counter MUST be used to acknowledge all the messages
>> up to that sequence number which have been sent."
>>
>> But the way I interpret that it requires that each reliable message
>> MUST be individually acknowledged.
>
> Not the way I read it, here's the section on ACKs
>
> When the following messages are received, an ACK MUST be sent in
> return: NEW, HANGUP, REJECT, ACCEPT, PONG, AUTHREP, REGREL, REGACK,
> REGREJ, TXREL.  ACKs SHOULD not be expected by any peer and their
> purpose is purely to force the transport layer to be up to date.
>
> Implicitly for any other FullFrame the ACK is optional, so the the CDR
> frame type would be able to make it's own rules.
>
> In implementation terms you have to be able to treat ACKs as optional
> because for some messages they are - NEW for example may be 'acked'
> with either an ACK, an ACCEPT a REJECT, AUTHREQ depending on the
> situation.
>
> (I admit, I'm cheating - I got the clue from a helpful comment from  
> Mark a couple of years back...)

I'm not picky about where enlightenment is obtained :-)

It does point out that some wordsmithing would be helpful.

If any of the authors are reading this could you take this as a request
for clarification on when ACKs are required?

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Re: [asterisk-dev] The New CDR system

2007-03-31 Thread Rod Dorman
On Saturday, March 31, 2007, 09:43:53, critch wrote:
> On Sat, 2007-03-31 at 10:59 +0300, Tzafrir Cohen wrote:
>>...
>> But CDR data really needs what TCP has to offer.
>> 
>> What you do is re-implement TCP (or TCP/with a bit of SSL). While it
>> might work, ,it is not as good as the original.
>
> The only thing needed is guarenteed delivery. IAX full frames in UDP
> already do that. Seems to work fine for most people.

I can understand Tzafrir's point of view. The IAX2 protocol draft states
 "Full frames are sent reliably, so all full frames require an
 immediate acknowledgment upon receipt"
whereas  TCP  utilizes a sliding window protocol that doesn't require an
ACK for every individual transmission.

 
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Re: [asterisk-dev] Clearing pick-up groups on Zap/ channels

2006-11-06 Thread Rod Dorman
On Monday, November 6, 2006, 15:17:18, Nic Bellamy wrote:
>   ...
> I've been trying to think of an easy, minimal-change way out of the
> zapata.conf inheritence problem (since it's not just pickupgroups that 
> have this behaviour, it's just worse with that since you can't reset it 
> at present).
>
> What about something simple like a "resetdefaults" item that will 
> restore all zapata.conf settings to hardcoded defaults, and clear out 
> the pickupgroup/callgroup stuff?
> Ie.
> pickupgroup=1
> callgroup=1
> channel => 1-3
> resetdefaults=yes ; likely need the =yes since it's key=value based
> otherconfig=123
> channel => 4

What about just setting it to an empty value?
e.g.
   pickupgroup =
 -or-
   pickupgroup = ""   (or some other empty value designation)

That way you could reset just the ones you want to.


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Re: [asterisk-dev] E&M + Dial Tone?

2006-06-18 Thread Rod Dorman
On Sunday, June 18, 2006, 12:54:44, Bart Fisher wrote:
> I ask on the user's newsgroup and got "dead air" - So trying here:
>   ...
> Any clues for me?

http://www.asterisk.org/support
Asterisk-users & Asterisk-dev
... asterisk-dev@lists.digium.com provides a forum for the
discussion of C code developement pertaining to the Asterisk project.
The asterisk-dev list is NOT a 2nd level support

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Re: [asterisk-dev] Writing a SIP client

2006-05-25 Thread Rod Dorman
On Thursday, May 25, 2006, 08:59:43, Vij wrote:
> I'm planning to write a sip client for windows using the sip code
> available in asterisk, so that when new features like JB are introduced to
> sip, they can be added seamlessly to the softphone too and would work with
> asterisk seamlessly. I plan to create a dll, so it would be easy for anyone
> to create a custom looking softphone with the all features of * sip code.

An informational RFC that just came out may be of interest to you

   RFC 4504
   Title:  SIP Telephony Device Requirements and Configuration
   Author: H. Sinnreich, Ed., S. Lass, C. Stredicke
   Status: Informational
   Date:   May 2006
   URL:http://www.rfc-editor.org/rfc/rfc4504.txt

   This document describes the requirements for SIP telephony devices,
   based on the deployment experience of large numbers of SIP phones and
   PC clients using different implementations in various networks. The
   objectives of the requirements are a well-defined set of
   interoperability and multi-vendor-supported core features, so as to
   enable similar ease of purchase, installation, and operation as found
   for PCs, PDAs, analog feature phones or mobile phones.

   We present a glossary of the most common settings and some of the
   more widely used values for some settings. This memo provides
   information for the Internet community.
   

If  nothing else you should browse section 3 "Glossary and Usage for the
Configuration Settings" for ideas on what to name your settings.

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Re: [asterisk-dev] Variable Inheritance, Setting Channel Variables outside of current context

2006-05-18 Thread Rod Dorman
On Thursday, May 18, 2006, 11:24:41, Johansson Olle E wrote:
> 18 maj 2006 kl. 16.41 skrev Peter Beckman:
>>  ...
>> I understand inheritance better now; there are child and parent
>> CHANNELS, and variables set in Channel A (parent) with a leading one
>> or two underscores will be inherited by Channel B (child) when
>> Channel A creates Channel B.
>
> Please don't use the phrase "parent" or "child" because there's no such
> relationship. When one channel creates another channel, we copy some
> data to the new channel. After that, it's two channels that may  
> become bridged.

But  clearly  there  *is*  a relationship at that point in time when the
creation  is  taking  place.  I  would  agree  that  parent/child  is  a
misleading  nomenclature  since other uses of it traditionally mean some
lasting association beyond the instant of creation.

If  labels  are  needed it would be good to be consistent. What would be
good labels to use? Some that come to mind are:
   creator/createe
   spawner/spawnee
   forker/forkee

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Re: [asterisk-dev] IAX internet draft (draft-guy-iax-00)

2006-03-06 Thread Rod Dorman
On Monday, March 6, 2006, 16:51:31, Steve Kann wrote:
>  ...
> I don't know if the spec should include this as a requirement or not:
> It was really a workaround for a broken implementation

IMHO  a  broken  implementation is equivalent to a different protocol. A
spec  should  describe how to do it right, not enumerate several ways to
do it wrong.

At  least  not in a normative section, a non-normative appendix would be
the appropriate way to incorporate it in the spec if its really needed.

> -- but it would still be needed for asterisk < 1.2, or very old
> iaxclient implementations (the new jitterbuffer was in iaxclient
> before asterisk).

I'm not familiar enough with IAX history. Were the older implementations
IAX Version 2?

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Re: [asterisk-dev] IAX internet draft (draft-guy-iax-00)

2006-03-06 Thread Rod Dorman
On Monday, March 6, 2006, 14:26:59, Steve Kann wrote:
> Adrian Sietsma wrote:
>>   ...
>> This would imply that _all_ frames received subsequently would be 
>> ignored, until frame #4 re-arrives, and resets the sequence ?
>
> That's correct.  That's how it works presently, and the way it would 
> have to work, unless the receiver stored out-of-order frames (which 
> would be a worthwhile optimization to make for the lossy link case, as 
> you note below).  I think that such an "out-of-order" reciever store 
> could be done without changing the specs, though -- as long as it 
> doesn't _act_ on frames out-of-order, it could probably defer processing
> them until it got the missing frame.

Is it worth stating that explicitly by saying something along the lines
of:
"The receiver MAY store out-of-order frames but MUST NOT process
them until it receives the missing frame."

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Re: [asterisk-dev] Zaptel 1.0 to latest version update

2006-02-07 Thread Rod Dorman
On Tuesday, February 7, 2006, 14:16:19, Kevin P. Fleming wrote:
> Rod Dorman wrote:
>> I  read  the  intent of the new Development and Release Cycle to be that
>> every x.y release will be "consumer ready".
>
> We are leaving the option open to make releases from the developer 
> branch as we get closer to making a real release. So, as opposed to 
> releasing 1.4.0-beta1, -beta2, -beta3, etc. we may release 1.3.90, 
> 1.3.91, 1.3.92, etc.

Ah, the first concrete answer, thanks. This is all I was asking about, I
didn't  mean  for  this  thread  to  wander  off  into  a  philosophical
discussion of release numbering :-)

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Re: [asterisk-dev] Zaptel 1.0 to latest version update

2006-02-07 Thread Rod Dorman
On Tuesday, February 7, 2006, 14:26:03, Tzafrir Cohen wrote:
> On Tue, Feb 07, 2006 at 01:57:32PM -0500, Rod Dorman wrote:
>>  ...
>> I  read  the  intent of the new Development and Release Cycle to be that
>> every x.y release will be "consumer ready".
>
> You can think of the current development branch as version 1.3.x . e.g:
> 1.3.0. . If you really need the version formatted as a version
> number.

I don't think that the current development branch really needs a version
number, its simply the current development branch aka trunk aka head.

Its the "release" version numbering I was asking about.

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Re: [asterisk-dev] Asterisk TCP

2006-02-05 Thread Rod Dorman
On Sunday, February 5, 2006, 06:18:01, asterisk_dev wrote:
> "Olle E Johansson" wrote:
>>  ...
>> The TCP implementation requires quite a lot of work to be done
>> properly and that patch implements some, but very little of what is
>> required. SIP over TCP needs the new socket interface that Kevin has
>> on hist to-do list and a serious change of chan_sip internals, which
>> is on my to-do list under the project name "chan_sip3".
>
> Great, so, how can I start? I think a good start would be looking at your
> chan_sip3. I've never been in an open-source project, so I'm a little lost
> with that, it would be nice if someone can help me with that.

You can view the code at
http://svn.digium.com/view/asterisk/team/oej/chan_sip3/


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Re: [Asterisk-Dev] Re: cvs revision tags

2005-11-21 Thread Rod Dorman
On Monday, November 21, 2005, 03:59:49, Olle E. Johansson wrote:
> Tony Mountifield wrote:
>>   ...
>> Yes, that's v1-2. It will only have bug fixes, and not experimental or new
>> features. The latter will continue to go into HEAD.
>
> HEAD that is now version 1.3dev, open for a lot of new crazy stuff and
> still not recommended for production use. The amount of crazyness will
> increase during the coming months so be prepared for the worst and stay
> on the v1-2 track for production servers.

I  was  tracking  HEAD  whilst  initially playing with Asterisk and will
probably  do  so again when I can scrape up the pieces to put together a
test  box.  I'm  definitely NOT tempted to track HEAD now that we've cut
over to Asterisk :-)

> Also remember that the v1-2 cvs branch is *not* the release. The release
> that we consider stable is the actual tarballs or the tagged CVS, like
> v1-2-0. The CVS will change while we are testing bug fixes and you may
> check out something we haven't tested out fully yet, which may cripple
> your stable server. CVS branches are always changing and committers may
> make mistakes.

Yes, I understand now that the v1-2-0 tag is the equivalent of burning a
CD, its a point in time and is not going to change.

However,  bugs  will  be discovered and changes made to HEAD to fix them
along  with  changes for the "new crazy stuff". I'm trying to figure out
what  the  best  approach  to  getting  those 1.2.0 'fixes' applied to a
production system is.

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Re: [Asterisk-Dev] Asterisk or Polycom Bug? - Message waiting

2005-11-08 Thread Rod Dorman
On Monday, November 7, 2005, 18:25:08, Will McCown wrote:
>   ...
> The bad news it that the light remains lit even if there are no
> new messages (as I pretty much expected).  So, it appears that if
> the polycom gets a message-summary with "Messages-Waiting: no"
> then it ignores the rest of the message and sets all of the counts
> to zero.  Reading the RFC I can see how this might be considered
> the correct behavior.

My  take  on RFC 3842 is that Messages-Waiting: is a 'new' messages flag
only  and  the  presence of 'old' messages are only communicated via the
xxx-Message: lines.

In particular "3.11. Rate of Notifications" where it has the phrase
   " ...which do not affect the overall message waiting state
(e.g., there are still new messages)."

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