[asterisk-dev] New Digium Employee, Terry Wilson
Greetings, Please join me in welcoming Terry Wilson to the Software Engineering team of Digium. Terry has been an active member of the Asterisk community for years, and has contributed to Asterisk development in many important areas. He has worked closely with other Asterisk developers in the past and was present at the developer conference this past May. We are excited to have him on board and look forward to his contributions helping us continue to improve Asterisk. Welcome! -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample
Mark Michelson wrote: > I just checked, and in 1.4 and trunk, static members, realtime members, > manager-added members, CLI-added members, and dialplan-added members > will all have their membername set to their interface if no membername > is specified. Yep, I see that now. I'd prefer it be done at the lowest level (create_member) instead of in the higher places that end up calling create_member, but what we have now works. Given that, I've removed the conditional logic that I put in yesterday since it isn't necessary. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] ast_readaudio_callback() called just one time
Hi,
I have the following behavior (Asterisk 1.2.24):
Using my channel driver, If my dialplan starts directly with a playback
functions, for example:
exten => 4020,1,VoicemailMain(${CALLERID(num)[EMAIL PROTECTED]);
the first messages wasn't played (in this case vm-login). The vm-password
file is played normally.
However, If I insert a Wait() function before:
exten => 4020,1,Wait(1)
exten => 4020,1,VoicemailMain(${CALLERID(num)[EMAIL PROTECTED]);
both vm-login and vm-password are played normally.
In the first case (without Wait()), checking the Asterisk code, I saw that
the ast_write() function is called just one time when the vm-login is
played. I found in file.c the ast_readaudio_callback() that should be called
several times until the file finishes. However, it is called just one time.
I checked that in my side, the audio is already enabled, the channel is UP
but my write callback is never called more than one (since ast_write() is
not called as well).
What is the difference using Wait or not? Why the vm-login fails to play in
the first case?
Any idea where to find the cause ?
Thanks!
--
--
Paulo Garcia
Pika Technologies Inc
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Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample
Mark Michelson wrote: > Correct me if I'm wrong, but I believe that the membername field of a > member will be set to it's interface if no name is specified. There may > be some exception, but I'm pretty certain this happens for all types of > members. I was just working in that code yesterday and didn't see that happening in create_member()... but it would be a good way to solve this problem (and eliminate a bunch of conditionals that already exist). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample
Mark Michelson wrote: > Kevin P. Fleming wrote: > >> SVN commits to the Asterisk project wrote: >> >> >> >>> + if (update_cdr && qe->chan->cdr) >>> + ast_copy_string(qe->chan->cdr->dstchannel, >>> member->membername, sizeof(qe->chan->cdr->dstchannel)); >>> >>> >> This is buggy; member->membername could be an empty string, it is only >> populated if a member name is specified. For static members there is no >> membername, and so if 'updatecdr' is enabled this will result in a >> broken CDR entry. >> >> This code either needs to use member->interface if member->membername is >> empty, or we need to simplify the code in app_queue and just populate >> ->membername with a copy of ->interface if no member name is provided >> when the member is added to the queue. >> >> >> > Correct me if I'm wrong, but I believe that the membername field of a > member will be set to it's interface if no name is specified. There may > be some exception, but I'm pretty certain this happens for all types of > members. > > Mark! > I just checked, and in 1.4 and trunk, static members, realtime members, manager-added members, CLI-added members, and dialplan-added members will all have their membername set to their interface if no membername is specified. Mark! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AST_FRAME_DIGITAL
Klaus Darilion wrote: > Thus, to summarize: > > I see two options for adding digital/3G video types to Asterisk: > > 1. AST_FORMAT_H223. This will only be used if the incoming call is > digital AND signals H.223&H.245. > > + with this option it is for sure that the digital call is a 3G video call > - there is no standardized way to forward this to other Asterisk server > - does not help us handling other non-3G-video calls > > 2. AST_FORMAT_CLEARMODE. This will be used for any incoming digital > call. Thus, from the frametype it is not deriveable which application is > inside the CLEARMODE. > > + by adding CLEARMODE to rtp.c we have a standard conform way to forward > digital calls to other Asterisk servers > + more generic and flexible > - requires a little bit more dialplan logic to detect 3G video calls > > Thus, is there a chance to get AST_FORMAT_CLEARMODE into Asterisk (which > I would prefer instead of AST_FORMAT_H223)? Although it probably sounds very attractive, functionally this is the same thing as the proposal to use AST_FRAME_DIGITAL as an opaque container. So if you were using AST_FORMAT_CLEARMODE, you are not going to be able to tell the other end of the IAX connection about what it is terminating, so if it needs to bridge the call to a non ISDN endpoint (such as a videophone) this will not be possible. AST_FORMAT_CLEARMODE is not the answer, at least for the H.223 and H.245 data problem that we are discussing. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample
Kevin P. Fleming wrote: > SVN commits to the Asterisk project wrote: > > >> +if (update_cdr && qe->chan->cdr) >> +ast_copy_string(qe->chan->cdr->dstchannel, >> member->membername, sizeof(qe->chan->cdr->dstchannel)); >> > > This is buggy; member->membername could be an empty string, it is only > populated if a member name is specified. For static members there is no > membername, and so if 'updatecdr' is enabled this will result in a > broken CDR entry. > > This code either needs to use member->interface if member->membername is > empty, or we need to simplify the code in app_queue and just populate > ->membername with a copy of ->interface if no member name is provided > when the member is added to the queue. > > Correct me if I'm wrong, but I believe that the membername field of a member will be set to it's interface if no name is specified. There may be some exception, but I'm pretty certain this happens for all types of members. Mark! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [svn-commits] rizzo: branch rizzo/video_v2 r82753 - /team/rizzo/video_v2/channels/
On Tue, Sep 18, 2007 at 11:06:12AM -0500, Jason Parker wrote: > SVN commits to the Digium repositories wrote: > > Author: rizzo > > Date: Tue Sep 18 10:45:22 2007 > > New Revision: 82753 > > > > URL: http://svn.digium.com/view/asterisk?view=rev&rev=82753 > > Log: > > lots of cleanup of this module, including the ability > > to run in "headless" configuration i.e. without X or SDL. > > > > aalib? :) it's actually not too unreal considering the geometry of videocall streams. I just need to figure out how to tell SDL to use aalib (or some other backend) instead of X11! cheers luigi ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] call for testers for team/rizzo/video_v2
Hi,
i have reached a hopefully reasonable state for the code in
http://svn.digium.com/view/asterisk/team/rizzo/video_v2/
so it would be good if someone could give it a try and send feedback.
In a nutshell, this branch lets you send and receive (and display)
H263+ video with chan_oss (tested) and chan_alsa (not well tested),
using either a webcab or an X11 window as source.
The requirements are SDL, a recent ffmpeg (July 2007), and optionally
Video4Linux v.1 to use the webcam.
The branch includes also updated config files (oss.conf, extension.conf,
sip.conf) so you should not have much trouble, and just run
console dial SIP/[EMAIL PROTECTED]
to get an audio/video call up and running.
I have developed and tested this on FreeBSD and chan_oss mostly,
(and a bit on chan_alsa), talking to another asterisk+chan_oss/FreeBSD,
linphone/linux, X-lite/Windows. The chan_alsa version is less stable.
I would be very grateful if someone could give this code a try and
send feedback to me (or to the list).
I have to say that the libraries used (ffmpeg and SDL) are extremely
fragile and it is likely to crash the program on unexpected input - this
includes programmer's errors and bogus data coming from the network.
Changes to the config files below, if you should decide to merge
them with your own configs.
thanks
luigi
oss.conf ---
; Additional settings to try videosupport.
; 'videodevice = /dev/video0' uses your V4L webcam as video source
; 'videodevice = X11' (capital) transmits the content of the upper left
; corner of your X11 display,
; videowitdh and videoheight set the resolution - please note that
; codecs and other videophones only support some standard resolutions
; e.g. 176x144, 352x288, 320x240 ...
; 'bitrate' is the net video bit rate (plus you have RTP/UDP/IP overhead,
; also consider that the traffic is very bursty
[general](+) ; additional settings for videosupport
overridecontext=yes; so you can specify a SIP/[EMAIL
PROTECTED] 'number'
videodevice = X11 ; X11 grabber
; videodevice = /dev/video0; Use this if you have a webcam...
videowidth = 352 ; either 352x2
videoheight = 288 ;
fps = 15 ;
bitrate = 65000;
sip.conf ---
[general](+) ; useful options for video support
videosupport = yes
allow = h263p
extensions.conf
[default](+); used for console
; the following is useful to dial a SIP url from the console, with
; "console dial SIP/[EMAIL PROTECTED]:portnumber"
; (remember to set "overridecontext=yes" in oss.conf
exten => _[sS][iI][pP].,1,Dial(${EXTEN:},,r)
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Re: [asterisk-dev] [asterisk-commits] qwell: trunk r82800 - in /trunk: apps/app_queue.c configs/queues.conf.sample
SVN commits to the Asterisk project wrote: > + if (update_cdr && qe->chan->cdr) > + ast_copy_string(qe->chan->cdr->dstchannel, > member->membername, sizeof(qe->chan->cdr->dstchannel)); This is buggy; member->membername could be an empty string, it is only populated if a member name is specified. For static members there is no membername, and so if 'updatecdr' is enabled this will result in a broken CDR entry. This code either needs to use member->interface if member->membername is empty, or we need to simplify the code in app_queue and just populate ->membername with a copy of ->interface if no member name is provided when the member is added to the queue. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [svn-commits] rizzo: branch rizzo/video_v2 r82753 - /team/rizzo/video_v2/channels/
SVN commits to the Digium repositories wrote: > Author: rizzo > Date: Tue Sep 18 10:45:22 2007 > New Revision: 82753 > > URL: http://svn.digium.com/view/asterisk?view=rev&rev=82753 > Log: > lots of cleanup of this module, including the ability > to run in "headless" configuration i.e. without X or SDL. > aalib? :) -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] nice Asterisk
Is anybody who can tell me, how about this linux commond nice -n work with asterisk? -- serva 2007-09-18 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Applicationmap and multiple Key/ActivatedBy entries.
On 9/17/07, Russell Bryant <[EMAIL PROTECTED]> wrote:
> Atis wrote:
> > [applicationmap]
> > nway_start_target => 00,self/callee,Macro,nway_start
> > nway_start_source => 00,self/caller,Macro,nway_start
>
>
>
> > if (!strcmp(feature->exten, code)) {
> > if (option_verbose > 2)
> > ast_verbose(VERBOSE_PREFIX_3 " Feature Found: %s exten:
> > %s\n",feature->sname, tok);
> > res = feature->operation(chan, peer, config, code, sense, feature);
> > AST_LIST_UNLOCK(&feature_list);
> > break;
> >
> > So, it just calls first feature and activatedby checking is done
> > inside. I would be willing to play with this, but my C experience is
> > quite limited. So, how should this be best done - create function for
> > checking that feature can be used, and do operation() only then. Or
> > check some result codes of operation() (for now i got 21 and 23 - they
> > are meaningless to me).
>
> I would consider this a valid bug. In the case that the feature doesn't
> execute
> because of the activated on/by setting, it should return a result that
> indicates
> that the code should keep trying other features for a match.
>
> I went ahead and just fixed it. See 1.4 and trunk revisions 82594 and 82595.
>
> Thank you for describing the problem extremely well!
Thanks, it seems working.
Regards,
Atis
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IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org
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