[asterisk-users] MOH realtime using only one thread for MOH
Hi, When using realtime MOH and an external script such as madplay if in MusicOnhold we have set cachertclasses=yes, after an Asterisk restart the first time a call comes in madplay is started. If ten people are using the same moh class there is only one instance of madplay. The issue is that if no one is using the MOH class once there is an instance of madplay it's up forever (till Asterisk is restarted). On the other hand if cachertclasses is set to no then it will start up madplay when the MOH class is used and it will stop when the call is done. The issue with this is if there are 10 people using this MOH class then there are ten instances of madplay being used. Is there any way of having a happy mix of both? If the calls is not in use then launch an instance (as it is with cachertclasses=yes) however when the MOH is over if there is no one in the class to stop the instance of madplay running (as it is if we have cachertclasses=no). TIA. Dovid -- _ -- 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] [CFP] reminder! FOSDEM RTC dev-room talks: deadline tomorrow
Reminder: speaker's deadline tomorrow, 17 November at 23:59 UTC The Free RTC dev-room has already received some really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the original call for participation: https://danielpocock.com/fosdem-2017-rtc-cfp We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc -- _ -- 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] dahdi_scan
On Mon, Nov 14, 2016 at 07:36:40AM -0500, Jerry Geis wrote: > >dahdi_scan (and lsdahdi) shows data reported by the DAHDI kernel modules > >themselves. lspci shows the PCI device. > > >What is the output from dahdi_hardware ? It should show if there's a > >module handling this device, and also which module. > > > lsdahdi > > lspci | grep Digium > 03:05.0 Ethernet controller: Digium, Inc. Wildcard TE122 single-span > T1/E1/J1 card (rev 11) > > dahdi_hardware > pci::03:05.0 wcte12xp-d161:8001 Wildcard TE122 So as you can see, there's no kernel module that handles this device. Any change if you run: modprobe wcte12xp #? Do you have dahdi-linux installed? Properly? If you think it is, what is the output of: lsmod | grep dahdi modinfo dahdi uname -r find /lib/modules/`uname -r` -name dahdi.ko find /lib/modules -name dahdi.ko -- 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] iaxmodem errors.
Hi. the fax show version does not work since i am not using the digium
modem.
the iax2 show peers is the command for me and the output is:
PBX*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Description
iaxmodem/iaxmod 127.0.0.1 (S) 255.255.255.255 4570 OK
(1 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
PBX*CLI>
the problem is that in logs i am getting errors and i do not know how to
fix it.
root@PBX: /var/log/iaxmodem $ more ttyIAX0
[2016-11-16 09:08:12.483144] Registration failed.
[2016-11-16 09:13:05.118692] Terminating on signal 15...
[2016-11-16 09:21:49.181872] Registration failed.
[2016-11-16 09:22:30.731893] Terminating on signal 15...
[2016-11-16 09:22:30.759221] Registration failed.
[2016-11-16 09:25:11.014642] Registration failed.
root@PBX: /var/log/iaxmodem $
Any ideas?
On 15/11/2016 5:40 μμ, Larry Moore wrote:
I suspect I followed a guide much like the one you have used including
information found on voip-info - sorry, I can't seem to find any
bookmarks of relevant information.
I spent an enormous amount of time getting it working and working very
well, the real issue was getting T.38 working - I applied a patch to
Asterisk version 1.8 to get the T.38 gateway functionality.
I would have started off my testing by confirming communications
between two IAX modems, I presume you are using HylaFAX too.
Once the communications between the two IAX modems was working I
progressed with testing sending and receiving faxes using G711A
through my VoIP service and a modem attached to a PSTN service,
suffice to say T.38 functionality was the key to getting reliable
faxes working through VoIP at least when traversing the Internet,
fortunately my VoIP provider facilitates T.38.
Using an SPA8800 on my network I tested sending and receiving faxes
with a modem attached to the SPA8800, it worked in G711A and T.38.
I progressed to Asterisk 11 where the T.38 gateway functionality is
better along with other improvements.
What is the output on your system for:
fax show version
Cheers,
Larry.
On 15/11/2016 8:09 PM, tux john wrote:
Hi. Since I am messing a lot with it without seeing the end of, may I
ask if there is any solid guide for that please?
On 13/11/2016, 07:42 Larry Moore wrote:
Some additional information which may help you with your
installation.
I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use iaxmodem3
for outbound fax transmissions.
I created a queue for the other 3 modems, here is my entry in
queues.conf:
[hylafax-iax]
strategy=linear
ringinuse=yes
autopause=no
retry=4
timeout=5
timeoutpriority=conf
reportholdtime=no
joinempty=strict
leavewhenempty=strict
musicclass=silence
member => IAX2/iaxmodem2
member => IAX2/iaxmodem1
member => IAX2/iaxmodem0
In case you are wondering about the 'musicclass' I have used,
here is the section from musiconhold.conf, the actual location of
the files may be elsewhere on your system:
[silence]
mode=files
directory=/usr/local/share/asterisk/silence
; ls /usr/local/share/asterisk/silence
; 10.gsm
;
; The file 10.gsm came from
/usr/local/share/asterisk/sounds/en/silence
I changed 'callbackextension' in my sip.conf for the trunk so
that it would go directly to the 'fax' extension in the dialplan
i.e. 'callbackextension=fax'.
I've included the console output when an incoming fax is received:
== Using SIP RTP TOS bits 184
-- Executing [fax@from-itsp:1] NoOp("SIP/itsp-0044",
"Fax Detected 2016-11-13 12:33:40 +0800") in new stack
-- Executing [fax@from-itsp:2]
GotoIf("SIP/itsp-0044", "0?3:8") in new stack
-- Goto (from-itsp,fax,8)
-- Executing [fax@from-itsp:8] NoOp("SIP/itsp-0044",
"Finish if_from-itsp_237") in new stack
-- Executing [fax@from-itsp:9]
GotoIf("SIP/itsp-0044", "0?10:13") in new stack
-- Goto (from-itsp,fax,13)
-- Executing [fax@from-itsp:13] NoOp("SIP/itsp-0044",
"Finish if_from-itsp_238") in new stack
-- Executing [fax@from-itsp:14] Set("SIP/itsp-0044",
"FAXOPT(gateway)=yes") in new stack
-- Executing [fax@from-itsp:15]
Queue("SIP/itsp-0044", "hylafax-iax,dRt,,,15") in new stack
-- Started music on hold, class 'silence', on
SIP/itsp-0044
-- Call accepted by 127.0.0.1 (format alaw)
-- Format for call is (alaw)
-- IAX2/iaxmodem2-3086 is ringing
-- Stopped music on hold on SIP/itsp-0044
-- IAX2/iaxmodem2-3086 answered SIP/itsp-0044
> 0x89bac000 -- Probation passed - setting RTP source
address to
