Re: [asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1
>Hi Jerry, > just had a look through the code, and from what I can tell, what >you're trying to do is not supposed to work, exactly. It appears that >what Asterisk expects is to be given a filename, such as "myplayback". >Asterisk will first search for an audio version of the file (like >myplayback.gsm or myplayback.opus), and open that as an audio stream. If >that succeeds, it then will also see if there is an accompanying video >stream (such as myplayback.h264). If it then finds that video, then the >result will be that Asterisk will play the audio from the audio file and >the video from the video file. >What this means is that Asterisk does not properly handle: >* Files that have audio and video streams contained within >* Video files without accompanying audio >This is one of those times where Asterisk's handling of video is not >user-friendly and in general ass-backwards and terrible. If you have a >tool that can extract the audio to its own file, then you would be able >to run your scenario, presumably. >It would be a welcome addition for Asterisk to be able to open a single >file containing video and accompanying audio and be able to play those back. Hi Mark, Thanks for your reply... I just tried what you suggested on only got audio. I created a wav file and put it in the /tmp directory just like the video.h264 file. So /tmp has video.h264 and video.wav both. I then placed the call and only heard the audio from the wav file. I used this for my call file: Channel: SIP/2002 Context: testing Extension: 99 Priority: 1 Application: Playback Codecs: h263,h264,vp8,g722,ulaw,alaw,wav Data: /tmp/video My Bria 4 softphone uses the h263 and h264 codecs and of course wav file audio. Based on your look of the code did I miss something to trigger the playing of the video file? I can extract the audio out to a seperate file - so not a show stopper for me. No errors showed up on the Asterisk CLI when I did my test. Thanks so much, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1
On 12/19/2016 07:39 PM, Jerry Geis wrote:
I can create an audio call file and specify Application: Playback and Data: a
path to the audio file, it calls the phone and plays the audio message just
fine.
I am trying to do the same with a video file. I specify Application: Playback
and Data: the path to the video file (no ending of course), and I do specify
also the Codecs: h264,h263 etc...
Asterisk reports:
*File /tmp/video does not exist in any format
*>* Unable to open /tmp/video (format ulaw|h263|h264)*
Looking then at the code and attaching with the debugger.
the ast_openstream_full() function has this condition:
if (!fileexists_core(filename, NULL, preflang, buf, buflen,
file_fmt_cap) || !ast_format_cap_has_type(file_fmt_cap, AST_MEDIA_TYPE_AUDIO)) {
}
So fileexists_core() returns 1 but the next call to ast_format_cap_has_type()
fails. because its looking for AST_MEDIA_TYPE_AUDIO and the file is a video
file. Nowhere is the an AST_MEDIA_TYPE_VIDEO.
I can use the call file to setup a video call between two video softphones just
fine. However using the call file to call a phone and play a video is not
working at all for me.
Am I on the right track?
Is this supposed to work? if so how since there is no check of the
AST_MEDIA_TYPE_VIDEO?
Thanks,
Jerry
Hi Jerry,
I just had a look through the code, and from what I can tell, what
you're trying to do is not supposed to work, exactly. It appears that
what Asterisk expects is to be given a filename, such as "myplayback".
Asterisk will first search for an audio version of the file (like
myplayback.gsm or myplayback.opus), and open that as an audio stream. If
that succeeds, it then will also see if there is an accompanying video
stream (such as myplayback.h264). If it then finds that video, then the
result will be that Asterisk will play the audio from the audio file and
the video from the video file.
What this means is that Asterisk does not properly handle:
* Files that have audio and video streams contained within
* Video files without accompanying audio
This is one of those times where Asterisk's handling of video is not
user-friendly and in general ass-backwards and terrible. If you have a
tool that can extract the audio to its own file, then you would be able
to run your scenario, presumably.
It would be a welcome addition for Asterisk to be able to open a single
file containing video and accompanying audio and be able to play those back.
Mark Michelson
--
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[asterisk-users] [OT] Outbound on SPA3102 FXO stopped to work. Where to look at ?
Hello, I'm currently facing a quite strange issue. On a customer location, an old SPA3102 suddenly stopped to work a couple of days ago. More precisely, calls still come in but I can't dial out to PSTN. This box worked for several years for oubound and inbound calling. My setup is: Asterisk 11 <--SIP--> SPA3102 <---FXS/FXO---> Router with FXS port < @ ---> PTSN It took me days, long time ago, to adapt default SPA3102 config to local conditions. The box autoprovision itself from HTTP server. I didn't change anything lately in its config file. When I directly plug an analog into my router box, I can dial out. Symptoms are: I can't hear dialed digits anymore destination phone doesn't ring, SPA3102 Info page displays dialed number. Either, my provider changed something which bothers my SPA3102 somehow but not an analog phone, my SPA3102 "is getting old" and needs to be reconfigured a bit to behave as usual. Has anyone met something like this ? Suggestions ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax faling on PJSip
I am working on moving from version 11 to version 13 for my fax applications. We are bumping into an issue where the bulk of the T38 faxes are failing. The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE These same faxes succeed on the 11 version of asterisk. I am wondering if there are any ideas? COMREC_ERR_TRANSMIT_PHASE Both servers are running the same version of spandsp. The dialplan code is the same on both. The only difference is the versions of asterisk and pjsip on the 13 platform. Any ideas would be appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]
On Mon, Dec 19, 2016 at 05:10:42PM +0100, Olivier wrote: > Thanks for the tip: > changing to permissive mode made it ! > > Using methods suggested in [1], do you think its possible and worth the > effort to configure SELinux to work with Asterisk/Systemd in Enforcing mode > ? > > [1] https://wiki.centos.org/HowTos/SELinux I think it should be possible. IIRC I once gave it a shot and was mildly successful, but eventually gave up due to issues related to interaction with Apache. If you do run into a problem, I wonder what it is. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd
On Mon, Dec 19, 2016 at 03:54:47PM +0100, Olivier wrote: > Hello, > > For a new project, I'm adapting existing installation script to CentOS 7. > I must admit I don't understand how to adapt things to systemd. > > Here are my questions: > > 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib. > Do you think such directory and matching Makefile target could be useful ? > > 2. Should /run/asterisk directory creation be left to systemd or done by > installation script before running "systemctl start asterisk" ? > > 3. I edited the following /etc/systemd/system:asterisk.service file: > [Unit] > Description=Asterisk PBX and telephony daemon. > After=network.target > > [Service] > Type=forking > PIDFile=/var/run/asterisk/asterisk.pid Remove those two (or get latest version with sd_notify support, make sure it works, and use 'Type=notify') > Environment=HOME=/var/lib/asterisk > WorkingDirectory=/var/lib/asterisk > ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C Drop -F as well > /etc/asterisk/asterisk.conf > #ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf > ExecStop=/usr/sbin/asterisk -rx 'core stop now' I'm trying to think if this is needed. Anything wrong with just letting systemd kill asterisk and all of its child precesses? > ExecReload=/usr/sbin/asterisk -rx 'core reload' Also, IIRC: User=asterisk -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
