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 -
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On 20/03/2015 20:00, asterisk-users-request at lists.digium.com
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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 at gmail.com>
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] PJSIP Video on WebRTC Ast 13
> Message-ID: <9CE929C6-8E20-4794-A44F-E55AC877DAE7 at 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 at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
> Message-ID:
> <CAL_GE3To07V8gZ6SaCFhO1=x1JakTO595kCTMNLNkAaa-BqvTA at
mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> 2015-03-18 12:54 GMT-06:00 ricky gutierrez <xserverlinux at
gmail.com>:
>
>> 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
>
>
>
