similar to: Add & play moh-files without reload

Hello I did not mention it but of course the MOH directory is listed in /etc/asterisk/musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh [myfolder_1] mode=files directory=/var/lib/asterisk/moh/myfolder/1 sort=alpha [myfolder_2] mode=files directory=/var/lib/asterisk/moh/myfolder/2 sort=alpha [myfolder_3] mode=files directory=/var/lib/asterisk/moh/myfolder/3 sort=alpha

Le 23/03/2017 ? 20:17, Jonas Kellens a ?crit : > Hello > > > is there any more information on how to reload/read musiconhold files ? CLI> module reload res_musiconhold -- Daniel > On 07-03-17 10:46, Jonas Kellens wrote: >> Hello >> >> I did not mention it but of course the MOH directory is listed in >> /etc/asterisk/musiconhold.conf : >> >>

Hello using Asterisk 1.8.32.3 Current music on hold : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity New musiconhold

> Hello > as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. > Only at restart the new files are available. > Is this a bug ?? How can I get more debugging for this problem ?? I think there is currently a bug with MOH. For now, if you add a file to a moh folder, 'touch musiconhold.conf' and then reload moh.

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [root at asterisk testing]# asterisk -rx "file convert testExtended2.wav testExtended2.alaw" Unable to open input file: testExtended2.wav [root at asterisk testing]# asterisk -rx "file convert testLong2.wav testLong2.alaw" Unable to open input file: testLong2.wav The wav-file is MONO,

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , what is the meaning of the fields : `*digit*` char(1) NOT NULL default '', `*sort*` varchar(16) NOT NULL default '', and what are there default values ?! What is the default value of :

Hi list. Does anyone know how to keep calls between 2 local SIP-phones on the local private network when the 2 local IP-phones are registered to an online public Asterisk-server ?? What network-element / router do I need to install to prevent the RTP-traffic from flowing via the internet ? Config : Asterisk --internet-- > router/firewall --> connected local IP-phones Internal call :

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

Is there a way to tell asterisk to beep every few seconds rather than play MOH. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the

Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip