[asterisk-dev] How can retrieve a held call ...

2006-05-03 Thread md 



Hello,
I need retreive a held call that is held during a 
consultation call and the consulted extension has hangup. The sip phone 
(Swissvoice IP10S) is with 2 lines: one in held state and the other 
in disconnecting tone. How can i stop the disconnecting tone and retrieve 
the held call without touch the phone? I need do that from manager 
api.
 
Thank you
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Video, anyone?

2006-05-03 Thread Russell Bryant 
Olle E Johansson wrote:
> In order to open up a forum for those of you that want to work with
> Video, SIP and Asterisk, I can
> set up a temporary mailing list for the AVTF - Asterisk Video Task Force!

What is wrong with just using this mailing list?

Russell
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Corydon76

2006-05-03 Thread Kevin P. Fleming 
Denis Smirnov wrote:

> If there is one bug report with patch that fix it.
> And there is another enchantment, that bigger then first patch, and fix
> first bug also, but has different purpose, is this duplicate bug?

Yes. If the second patch cannot be applied if the first one is applied
(because they are fixing the same bug), then it's likely they will be
considered to be duplicates.

This is why we strongly suggest to people _not_ to combine bug fixes and
new features in the same patch.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Re: Dropping incompatible voice frame

2006-05-03 Thread Martin Vít 

Joseph Rothstein wrote:

It appears that the messages get generated following a forward set on the
phones:
  

i've exactly same issue when forwarding on SIP

May  3 16:20:19 NOTICE[11150]: channel.c:1911 ast_read: Dropping 
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin 
since our native format has changed to alaw


no idea how to debug this


Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Executing
Goto("IAX2/RemoteServ-7", "sanset|959|1") in new stack
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Goto (sanset,959,1)
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Executing
Macro("IAX2/RemoteServ-7", "stdexten|sanset|de") in new stack
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Executing
Set("IAX2/RemoteServ-7", "LANGUAGE()=de") in new stack
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Executing
SetMusicOnHold("IAX2/RemoteServ-7", "sanset") in new stack
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Executing
Dial("IAX2/RemoteServ-7", "SIP/959|20") in new stack
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Called 959
Feb  6 12:45:19 VERBOSE[7884] logger.c: -- Got SIP response 302 "Moved
Temporarily" back from 62.245.163.127
Feb  6 12:45:19 VERBOSE[17902] logger.c: -- Now forwarding
IAX2/RemoteServ-7 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/959-8434)
Feb  6 12:45:19 VERBOSE[17905] logger.c: -- Executing
Macro("Local/[EMAIL PROTECTED],2", "stdexten|sanset|de") in new stack
Feb  6 12:45:19 VERBOSE[17905] logger.c: -- Executing
Set("Local/[EMAIL PROTECTED],2", "LANGUAGE()=de") in new stack
Feb  6 12:45:19 VERBOSE[17905] logger.c: -- Executing
SetMusicOnHold("Local/[EMAIL PROTECTED],2", "sanset") in new stack
Feb  6 12:45:19 VERBOSE[17905] logger.c: -- Executing
Dial("Local/[EMAIL PROTECTED],2", "SIP/944|20") in new stack
Feb  6 12:45:19 VERBOSE[17905] logger.c: -- Called 944
Feb  6 12:45:19 NOTICE[17905] channel.c: Dropping incompatible voice frame
on Local/[EMAIL PROTECTED],2 of format slin since our native format has
changed to alaw
Feb  6 12:45:19 NOTICE[17905] channel.c: Dropping incompatible voice frame
on Local/[EMAIL PROTECTED],2 of format slin since our native format has
changed to alaw
Feb  6 12:45:19 NOTICE[17905] channel.c: Dropping incompatible voice frame
on Local/[EMAIL PROTECTED],2 of format slin since our native format has
changed to alaw
Feb  6 12:45:19 NOTICE[17905] channel.c: Dropping incompatible voice frame
on Local/[EMAIL PROTECTED],2 of format slin since our native format has
changed to alaw

Anyone have any ideas?

Joe



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

  


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925 ..... ( Alex ) Corydon76 Issue Deleted: 0006920 ( Alex Smirnov )

2006-05-03 Thread Steven 
On Wed, 2006-05-03 at 17:27 +0400, Denis Smirnov wrote:
> On Wed, May 03, 2006 at 01:34:49PM +0400, Alex Smirnov ( DigitalXXI ) wrote:
> 
> ASD> 1) Absolutely agree, it's not a asterisk task to code audio files in
> ASD> different codecs, 
> 
> At now Astersisk _do_ this.

And so does soo many other software apps.

> ASD> easy to convert format is more practical , than
> ASD> amount of different codded files.
> 
> Only if you have supercomputer with one client :)

Hmm, maybe you need to get a modern computer. I don't expect there is
anyone here that will agree that a VoIP phone qualifies as a
supercomputer. 

> I doesn't know any codec that can be "easy" converted to G.729, G.723,
> speex, iLBC and GSM. Do you know that you need to avoid all codec
> conversion in high-volume installations, if you can? And if you can have
> voice prompt in all supported formats, this save your money?

Good codec convertion avoidance is to pass the conversion off to the
endpoints. 

> ASD> 2) What is REAL need to support such codding in * , when u can use
> ASD> any software to convert it to whatever u like ?
> ASD> ( something like split & rule :) )
> 
> I not understand your idea.

He was agreeing with me that there are other CLI apps to handle the job.

> ASD> 3) and just curios - any ex. for lib usability , except commercial
> ASD> software ?
> 
> With commercial software no lib usability, because this lib would be
> GPL'ed for all but Digium clients, that buy license.

he also wants another example of what you propose would be a use of the
library. So far no one has agreed that using the asterisk codecs and
formats are a good choice, and not enough of a reason to go through the
effort of a library. So we want more examples that might sway us.
-- 
Steven <[EMAIL PROTECTED]>

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Bridging two H324M calls

2006-05-03 Thread Klaus Darilion 
Please edit the Wiki page and attach it to the Wiki page. Then we can a 
single point of information


regards
klaus

zhuoqun Li wrote:

Hi Sergio,
Could you leave your email address here so I can email my trace files  
to you?


regards,
Zhuoqun


Message: 4
Date: Wed, 3 May 2006 08:45:04 +0200
From: Sergio Garc?a Murillo <[EMAIL PROTECTED]
>
Subject: RE: [asterisk-dev] Bridging two H324M calls
To: "Asterisk Developers Mailing List" <
asterisk-dev@lists.digium.com >
Message-ID:
<[EMAIL PROTECTED]
>
Content-Type: text/plain; charset="iso-8859-1"

Could you share the dumped files at least?
They would be very usefull..



From: [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
] On Behalf Of zhuoqun Li
Sent: martes, 02 de mayo de 2006 18:19
To: asterisk-dev@lists.digium.com 
Subject: Re: [asterisk-dev] Bridging two H324M calls




Hi Klaus,
to record a live video conversation, you just need to insert some
pieces of code into chan_zap.c, i.e. in the part where chan_zap do
native bridging:I inserted several lines (e.g. tmp = write(ftrace,
f->data, f->datalen); ) in line 3464 ( zt_bridge(), chan_zap.c).
BTW, I did the  H324M call briding in a v-1.2.4 Asterisk in the UK.

regards,
Zhuoqun Li






Date: Tue, 02 May 2006 11:17:14 +0200
From: Klaus Darilion < [EMAIL PROTECTED]
>
Subject: Re: [asterisk-dev] Bridging two H324M calls
To: Asterisk Developers Mailing List <
asterisk-dev@lists.digium.com >
Message-ID: <[EMAIL PROTECTED]
>
Content-Type: text/plain; charset=ISO-8859-1;
format=flowed

zhuoqun Li wrote:
>  Hi,
>  I have successfully bridged H324m calls through
Asterisk (configured
> with a ISDN BRI interface).
>  I have aslo dumped the live video conversation
into a binary file.
>  What I did is a "native channel bridge" and the
dump functions are
> inserted in the zt_bridge() in chan_zap.c.
>  Hope this helps...

Can you share your code? E.g. post it on
bugs.digium.com 

regards
klaus

>
>
>  regards,
>  Zhuoqun Li
>
>
>
> --
>
> Message: 4
> Date: Fri, 28 Apr 2006 08:41:24 +0200
> From: Sergio Garc?a Murillo <
[EMAIL PROTECTED] 
> mailto:[EMAIL PROTECTED]> > >>
> Subject: RE: [asterisk-dev] Bridging two H324M
calls
> To: "Asterisk Developers Mailing List" <
> asterisk-dev@lists.digium.com

 >>
> Message-ID:
><[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>
> Content-Type: text/plain;  
charset="iso-8859-1"

>
> Klaus Darilion wrote:
>  > Hi Sergio!
>  >
>  > I've done this once and it worked
(relaying). But I was not able to
>  > record the sessions. When I tried the
various "recording"
>  > applications the video call setup did not
worked anymore. Relaying
>  > was only successful when the bridging was
done directly on the ISDN
>  > card.
>  >
>  > I did this once with an old Asterisk
version. With newer Asterisk
>  > version relaying is not possible anymore,
as the zaptel code changes
>  > some call parameters (from data calls to
anything else ...).
>  >
>  > I tried to debug this once (message 0025307)
>  > http://bugs.digium.com/view.php?id=3891
> < ht

Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925 ..... ( Alex ) Corydon76 Issue Deleted: 0006920 ( Alex Smirnov )

2006-05-03 Thread Denis Smirnov 
On Wed, May 03, 2006 at 01:34:49PM +0400, Alex Smirnov ( DigitalXXI ) wrote:

ASD> 1) Absolutely agree, it's not a asterisk task to code audio files in
ASD> different codecs, 

At now Astersisk _do_ this.

ASD> easy to convert format is more practical , than
ASD> amount of different codded files.

Only if you have supercomputer with one client :)

I doesn't know any codec that can be "easy" converted to G.729, G.723,
speex, iLBC and GSM. Do you know that you need to avoid all codec
conversion in high-volume installations, if you can? And if you can have
voice prompt in all supported formats, this save your money?

ASD> 2) What is REAL need to support such codding in * , when u can use
ASD> any software to convert it to whatever u like ?
ASD> ( something like split & rule :) )

I not understand your idea.

ASD> 3) and just curios - any ex. for lib usability , except commercial
ASD> software ?

With commercial software no lib usability, because this lib would be
GPL'ed for all but Digium clients, that buy license.

-- 
JID: [EMAIL PROTECTED]
ICQ: 58417635 (please, use jabber, if you can)

http://freesource.info/

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Corydon76

2006-05-03 Thread Denis Smirnov 
On Tue, May 02, 2006 at 04:15:04PM -0500, Tilghman Lesher wrote:

TL> No, the only things that result in bad karma are:  1) opening bugs that
TL> are duplicates of existing bugs,

There is one bug, that _I_ think is not duplicate, but _you_ think are
duplicate.

If there is one bug report with patch that fix it.
And there is another enchantment, that bigger then first patch, and fix
first bug also, but has different purpose, is this duplicate bug?

I think, that bugs with different enchantment patches from different authors is
different bugs. You think that it's duplicates.

Who are right? I'm not find answer in bug guidlines.

TL> 2) opening bugs as MAJOR when they
TL> are not in the 1.2 series, 3) opening bugs as CRASH when they do not
TL> manifest themselves as a crash.  These are all contrary to the bug
TL> guidelines, which are freely available on the bugtracker.  There are
TL> other reasons as well, but they are all for actions that are agreed upon
TL> beforehand.  I have no power at this time to create new reasons for
TL> negative karma on the bugtracker.

-- 
JID: [EMAIL PROTECTED]
ICQ: 58417635 (please, use jabber, if you can)

http://freesource.info/

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Bridging two H324M calls

2006-05-03 Thread zhuoqun Li 
Hi Sergio,
Could you leave your email address here so I can email my trace files  to you?

regards,
ZhuoqunMessage: 4Date: Wed, 3 May 2006 08:45:04 +0200
From: Sergio Garc?a Murillo <[EMAIL PROTECTED]>Subject: RE: [asterisk-dev] Bridging two H324M callsTo: "Asterisk Developers Mailing List" <
asterisk-dev@lists.digium.com>Message-ID:<[EMAIL PROTECTED]
>Content-Type: text/plain; charset="iso-8859-1"Could you share the dumped files at least?They would be very usefull..From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of zhuoqun LiSent: martes, 02 de mayo de 2006 18:19To: 
asterisk-dev@lists.digium.comSubject: Re: [asterisk-dev] Bridging two H324M callsHi Klaus,to
record a live video conversation, you just need to insert some pieces
of code into chan_zap.c, i.e. in the part where chan_zap do native
bridging:I inserted several lines (e.g. tmp = write(ftrace, f->data,
f->datalen); ) in line 3464 ( zt_bridge(), chan_zap.c).BTW, I did the  H324M call briding in a v-1.2.4 Asterisk in the UK.regards,Zhuoqun LiDate:
Tue, 02 May 2006 11:17:14 +0200From:
Klaus Darilion < [EMAIL PROTECTED]>Subject:
Re: [asterisk-dev] Bridging two H324M callsTo:
Asterisk Developers Mailing List < asterisk-dev@lists.digium.com>Message-ID:
<[EMAIL PROTECTED]>Content-Type:
text/plain; charset=ISO-8859-1; format=flowedzhuoqun Li wrote:>  Hi,>  I
have successfully bridged H324m calls through Asterisk (configured> with a ISDN BRI interface).>  I
have aslo dumped the live video conversation into a binary file.>  What
I did is a "native channel bridge" and the dump functions are>
inserted in the zt_bridge() in chan_zap.c.>  Hope
this helps...Can
you share your code? E.g. post it on bugs.digium.comregardsklaus>>>  regards,
>  Zhuoqun Li>>>>
-->>
Message: 4>
Date: Fri, 28 Apr 2006 08:41:24 +0200>
From: Sergio Garc?a Murillo < [EMAIL PROTECTED]>
[EMAIL PROTECTED]
[EMAIL PROTECTED]> >>>
Subject: RE: [asterisk-dev] Bridging two H324M calls>
To: "Asterisk Developers Mailing List" <>
asterisk-dev@lists.digium.com asterisk-dev@lists.digium.com
>>>
Message-ID:><[EMAIL PROTECTED]>
[EMAIL PROTECTED]>>>
Content-Type: text/plain;  
charset="iso-8859-1">>
Klaus Darilion wrote:>  >
Hi Sergio!>  >>  >
I've done this once and it worked (relaying). But I was not able to>  >
record the sessions. When I tried the various "recording">  >
applications the video call setup did not worked anymore. Relaying>  >
was only successful when the bridging was done directly on the ISDN>  >
card.>  >>  >
I did this once with an old Asterisk version. With newer Asterisk>  >
version relaying is not possible anymore, as the zaptel code changes>  >
some call parameters (from data calls to anything else ...).>  >>  >
I tried to debug this once (message 0025307)>  >
http://bugs.digium.com/view.php?id=3891>
< http://bugs.digium.com/view.php?id=3891   >>  >
>  >
but did not received any help and could not solved it myself.>>
Could it be possible to modify the zapdump app in order to make to>
bridge two incoming calls through a pipe or socket?>
It's probably easier than bridging two channels through asterisk.>
And it would not affect the H324M as the master-slave determination>
is done in H245.>>
Best regards>
Sergio

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] 80ms RTP packets and G.729

2006-05-03 Thread Kevin P. Fleming 
Alistair Cunningham wrote:

> Would any of the Digium developers have an expected date for a G.729
> codec for an Asterisk version with the packetization code? Will it be
> before the next stable version?

We will finish up the module API changes before the end of the DevCon
next week, and then I will make new G.729 codec binaries.

> How much effort would it be to port the packetization code to Asterisk
> 1.2? If we did so, would the G.729 codec work with it?

Probably not a lot of effort, and yes.

> (BTW: There's a trivial typo "misstng" in the above error message in SVN
> trunk revision 24378)

It's gone in my working branch already :-)
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Video, anyone?

2006-05-03 Thread Paul Cadach 
Hi,

Olle E Johansson wrote:
[skipped]
> During my recent tests with video phones (thanks Grandstream and
> Foniris!) I have found out
> that we have a list of things to do. I have also found out that there
> are a lot of developers out there
> that have done it already - meetme with selectable video streams,
> chan_local with video and other
> patches that we need to incorporate into Asterisk. Smaller changes
> now, bigger changes for 1.6.

Don't forget H.323 also have generic support for video transmission... You can
include me into the AVTF list.


WBR,
Paul.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] 80ms RTP packets and G.729

2006-05-03 Thread Alistair Cunningham 
We have a customer who wants to run SIP with G.729 and 80ms RTP packets. 
They want to do this as they've been told by a 3rd party that this will 
allow them to keep bandwidth within 16Kbps per call, which is all they 
have available.


I see that there's a "packetization" option in Asterisk subversion. 
Alas, we can't use it as the G.729 codec doesn't run on this version:


 [codec_g729a.so]May  3 11:56:17 WARNING[31336]: loader.c:731 
__load_resource: misstng mod_data for codec_g729a.so


Would any of the Digium developers have an expected date for a G.729 
codec for an Asterisk version with the packetization code? Will it be 
before the next stable version?


How much effort would it be to port the packetization code to Asterisk 
1.2? If we did so, would the G.729 codec work with it?



(BTW: There's a trivial typo "misstng" in the above error message in SVN 
trunk revision 24378)


--
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] Video, anyone?

2006-05-03 Thread Olle E Johansson 

Friends,

During my recent tests with video phones (thanks Grandstream and  
Foniris!) I have found out
that we have a list of things to do. I have also found out that there  
are a lot of developers out there
that have done it already - meetme with selectable video streams,  
chan_local with video and other
patches that we need to incorporate into Asterisk. Smaller changes  
now, bigger changes for 1.6.


In order to open up a forum for those of you that want to work with  
Video, SIP and Asterisk, I can
set up a temporary mailing list for the AVTF - Asterisk Video Task  
Force!


I have no knowledge of video codecs, standards and other strange  
stuff, so I rely on you there.
I can manage a branch for this work and come with input from a SIP  
standpoint.


It is ok to discuss IAX2 and Video as well - does it work with the  
jitterbuffer, does trunking work,

any room for improvements?

Mail me *OFF LIST* and I'll add you to my temporary ad hoc mail list  
so we can start working

on this.

Welcome to the AVTF!

/O
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] Sip -> Zap(ETSI-ISDN) always generates 183 with SDP ?

2006-05-03 Thread Peter Childs 


gday guys (n'gals).

I have a third party SIP platform which generates outbound calls via
asterisk to ISDN (Australia - so thats ETSI ISDN).   This platform doesn't
really like inband signalling on outbound calls (ie getting 183's with SDP
-- its fine with 180 Ringing etc...)

Having had a bit of a silly time with the sip.conf variable
progressinband=never,no,yes (arg!) I dug a little deeper into the chan_sip
code.

It appears on a SIP->Zap call the ISDN channel is opened, and before you can
say 'boo' sip_write() in chan_sip is called this appears to occurs prior
to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..)

sip_write doesn't seem to care at all what progressinband is set to, and if
it gets a frame when the SIP channel is not in AST_STATE_UP it generates a
183 with SDP (then sets SIP_PROGRESS_SENT)

Does this behaviour seem strange?   I'm not really sure if this is a bug, a
'its just like that' thing, or something strange with our ISDN that is
unusual?

In an ideal world (for me anyway... *grin*) I would think that
progressinband=never (or even progressinband=no) would mean that 180
Ringing, 486 Busy etc would be used and 183 Session Progress with SDP would
not...

I have done some basic testing and if I patch as follows...

--start patch--

*** chan_sip.c.orig 2006-05-03 15:26:28.0 +1000
--- chan_sip.c  2006-05-03 19:59:04.0 +1000
***
*** 2548,2555 
if (p->rtp) {
/* If channel is not up, activate early
media session */
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
!   transmit_response_with_sdp(p, "183
Session Progress", &p->initreq, 0);
!   ast_set_flag(p, SIP_PROGRESS_SENT);
}
time(&p->lastrtptx);
res =  ast_rtp_write(p->rtp, frame);
--- 2548,2566 
if (p->rtp) {
/* If channel is not up, activate early
media session */
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
!   /* We have found that on outbound
ISDN calls via zap   */
!   /* frames start heading for SIP
before any signalling from */
!   /* the ISDN -- if progressinband=yes
then do early media otherwise */
!   /* supress the 183 Session Progress
with SDP   */
!
!   if(ast_test_flag(p,SIP_PROG_INBAND)
== SIP_PROG_INBAND_YES) {
!
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
!   ast_set_flag(p,
SIP_PROGRESS_SENT);
!   } else {
!   //ast_log(LOG_WARNING, "
***PJC*** Supress Early Media Frame\n" );
!   ast_mutex_unlock(&p->lock);
!   return 0;
!   }
}
time(&p->lastrtptx);
res =  ast_rtp_write(p->rtp, frame);

--end patch--

then I don't see the 183 Session Progress with SDP, and things 'appear' to
go ok for my basic calls, however I see a good splash of these type of
messages when the call goes from RINGING to UP...

May  3 19:22:22 DEBUG[12005] chan_zap.c: Write returned -1 (Resource
temporarily unavailable) on channel 1

I guess I'm not sure if this is a bug, a 'feature', or a limitation.

Any help or pointers would be great.  I'm happy to bugs.digium.com it, but I
wanted to make sure I wasn't wasting anyones time.

For what its worth I'm using a plain Dial() (no 'r'), progressinband=never,
and the latest 1.2 from SVN for this testing.

Cheers,
  Peter


--
 Peter Childs
 NEC Business Solutions Ltd
 Ph:61-8-8301-4908 Mb:61-4-0819-7693
 IM: pjcinaus (yahoo)

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Corydon76 Issue De leted: 0006925 ..... ( Alex )

2006-05-03 Thread Alex Smirnov ( DigitalXXI ) 

adrldc> Message: 6
adrldc> Date: Tue, 02 May 2006 13:17:14 -0500
adrldc> From: Steven Critchfield <[EMAIL PROTECTED]>
adrldc> Subject: Re: [asterisk-dev] Corydon76  Issue Deleted: 0006925,
adrldc> 04-28-0617:49 Corydon76 Issue Deleted: 0006920
adrldc> To: Asterisk Developers Mailing List 
adrldc> Message-ID: <[EMAIL PROTECTED]>
adrldc> Content-Type: text/plain

adrldc> On Tue, 2006-05-02 at 21:19 +0400, Denis Smirnov wrote:
>> On Tue, May 02, 2006 at 11:43:24AM -0500, Steven wrote:
>> >> After moving some code to library it would be easy to write some code that
>> >> use asterisk modules. Like format convertors.
>> S> With the exception of some codecs that are patent encumbered, you should
>> S> be using SOX for your format conversions. Don't reinvent the wheel just
>> S> to make a stretched point.
>> 
>> wav file created by asterisk not wav file created by sox (look to GAIN in
>> format_wav.c).

adrldc> Doesn't matter who creates the wav file, it is a standard file format.
adrldc> As for GAIN, go look at the man page for sox, it supports gain via the
adrldc> vol function.

>> sox doesn't support G.729, G.723.1. sox doesn't support G.722.

adrldc> I know G.729 is patent encumbered, and G.723.1 I think is, So we
adrldc> shouldn't be too concerned about them. And as for G.722, why not get
adrldc> that built ito sox? I'm sure the sox group will love the additional
adrldc> codecs.

>> It would be easer to use codecs/formats in asterisk, for creating/reading
>> files created by asterisk.

adrldc> Formats, no. Codecs maybe. There is nothing special about the formats in
adrldc> asterisk. If there was something special, then they would be useless
adrldc> outside of asterisk. 

>> And G.729/G.723.1 support is needed.

adrldc> Fine, you are still probably better off starting with sox with all the
adrldc> filtering built in and more codecs and all.

 1 ) Absolutely agree, it's not a asterisk task to code audio files in
 different codecs, easy to convert format is more practical , than
 amount of different codded files.

 2) What is REAL need to support such codding in * , when u can use
 any software to convert it to whatever u like ?
 ( something like split & rule :) )

 3) and just curios - any ex. for lib usability , except commercial
 software ?


-- 
Best regards,
 Alexmailto:[EMAIL PROTECTED]

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] Corydon76

2006-05-03 Thread Paul Cadach 
Hello,

Koopmann, Jan-Peter wrote:
> It is one thing to say "Hey this does not work, it is a bug" and another to
say "Hey this does not work as expected, I created a small enhancement, see the
patch attached" like it was in this case.

[skipped]

You forget to remember about just extending support of some sort of existing
specifications (for example, facilities in PRI, etc.) which is not usually
required to be used by every user but just better to have been implemented
without waiting for misworking reports.

Sortly, if feature/enhancement is not required by every user, it's not a case to
close/ignore the ticket with good working patch. Elsewhere someone needs to
implement it again some time later...


WBR,
Paul.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev