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anicet.lanjaniaina at ...
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 Posted: Tue Mar 31, 2015 7:07 am Post subject: [asterisk-users] help : annoucement queue |
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Hi everybody,
I've a matter with the queue annoucement with the "thereare", because if
I put just one member in my configuration (member => SIP/2098), the ivr
gave me that I was the firt or second in the next at the queue. But the
problem is, if I add one member (eg: member => SIP/2098 and member =>
SIP/2099), the ivr don't gave me the range but It play the background
sound that I declare in my musiconhold.
Very thanks for your helps.
Have a nice day.
--
Anicet LANJANIAINA
Gulfsat Madagascar
(+261) 345 600 259
Service Technique -Blueline Madagascar www.blueline.mg -
Facebook : blueline Madagascar – Twitter : blueline_MG
Please think about the environment before printing this e-mail.
On 20/03/2015 20:00, asterisk-users-request@lists.digium.com wrote:
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Today's Topics:
1. PJSIP Video on WebRTC Ast 13 (Gosmac)
2. Re: res_xmpp.c:3468 xmpp_client_reconnect: (ricky gutierrez)
3. Re: Asterisk 13 : SILK codec ? (Steve Murphy)
4. Re: Asterisk switching bridge to native_rtp even with
direct_media=no (Matthew Jordan)
5. Re: Asterisk 13 : SILK codec ? (Matthew Jordan)
6. Problems playing an audio file over an intercom/paging system
(Tech Support)
7. Asterisk on OpenWrt (first time user) (Sebastian Kemper)
8. Dahdi ISDN logging (Grant Bagdasarian)
9. Re: Dahdi ISDN logging (Tony Mountifield)
10. UNREACHABLE peer (thufir)
11. Re: UNREACHABLE peer (dotnetdub)
12. Re: UNREACHABLE peer (thufir)
13. Re: UNREACHABLE peer (thufir)
14. Re: UNREACHABLE peer (thufir)
15. Re: Caller ID Names (Jordan Cook - Gyron Networks)
16. Re: Caller ID Names (Jordan Cook - Gyron Networks)
----------------------------------------------------------------------
Message: 1
Date: Thu, 19 Mar 2015 12:36:54 -0430
From: Gosmac <goseeped@gmail.com>
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PJSIP Video on WebRTC Ast 13
Message-ID: <9CE929C6-8E20-4794-A44F-E55AC877DAE7@gmail.com>
Content-Type: text/plain; charset=utf-8
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams  it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test .
i have two questions and i hope you could give me some advise.
1) after marking video packet I?m able to make Dial() between two webrtc peers but i get one way audio and video on callee party, ?after 3 minutes on call? i get two way audio and video on all parties seems to be not just a problem on a missing keyframe.
1.1) the 3 minutes delay only happen using chrome stable , could be a dtls problem when asterisk make an offer to other endpoint?
1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call.
2) after marking video packets i realize that when you make a call with video and you involve on dialplan an application like playback or music on hold any application that played audio files (audio and video never work).
2.1) asterisk is muggling the audio and video streams ?
This is good information for all guys out there that wants to support video on webrtc in asterisk 13
Javier Riveros
------------------------------
Message: 2
Date: Thu, 19 Mar 2015 11:42:36 -0600
From: ricky gutierrez <xserverlinux@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
Message-ID:
<CAL_GE3To07V8gZ6SaCFhO1=x1JakTO595kCTMNLNkAaa-BqvTA@mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
2015-03-18 12:54 GMT-06:00 ricky gutierrez <xserverlinux@gmail.com>:
| Quote: | I'm confused this is not a patch, it's just garbage  , I'm making a
connection xmpp with asterisk and not connected, at the cli shows me
the message every second:
RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection
available when trying to connect client '
RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection
available when trying to connect client '
RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection
available when trying to connect client '
[2015-03-18 12:53:49] ERROR[2545]: res_xmpp.c:3468
xmpp_client_reconnect: No XMPP connection available when trying to
I hope not bother to write directly matt
regardss
|
Hi , any help , any info?
regardss
|
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anicet.lanjaniaina at ...
Guest
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| Posted: Wed Apr 01, 2015 12:18 am Post subject: [asterisk-users] help : annoucement queue |
|
|
Hi everybody,
I've a matter with the queue annoucement with the "thereare", because if
I put just one member in my configuration (member => SIP/2098), the ivr
gave me that I was the firt or second in the next at the queue. But the
problem is, if I add one member (eg: member => SIP/2098 and member =>
SIP/2099), the ivr don't gave me the range but It play the background
sound that I declare in my musiconhold.
ipbx-digue*CLI> core show version
Asterisk 1.8.13.1~dfsg1-3+deb7u3 built by pbuilder @ pungenday on a
x86_64 running Linux on 2014-01-04 01:03:48 UTC
Very thanks for your helps.
Have a nice day.
--
--
Anicet LANJANIAINA
Gulfsat Madagascar
(+261) 345 600 259
Service Technique -Blueline Madagascar www.blueline.mg -
Facebook : blueline Madagascar – Twitter : blueline_MG
Please think about the environment before printing this e-mail.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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