similar to: 1.6.1: iax trunk needs "dahdi timing" ??

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

>Information on timing sources can be found here: >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces >As noted on that page, ConfBridge can use any timing interface Asterisk >provides, and is not limited to the DAHDI timing interface. Generally, >timerfd is a good timing interface. >That aside, I would try to rule out external issues with the garbled audio

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

Folks, I've read some sources claiming that Asterisk does not need DAHDI for timing in 1.6.1. Is this true? Searching the web, all I can find is pages celebrating the fact but no actual documentation on which version it was introduced in and how one would go about configuring an external time source. I'm having a devil of a job trying to compile DAHDI on a hosted Xen VM and thought I

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Hi all I am using asterisk 11.24.1 on a centos 5 machine. kernel 2.6.18 flavor. (x86_64). I have about SIP 150 endpoints on it. when I send a message I'm getting garbled audio. I used to have a single PRI card in the box - but something happened and that connection no longer worked. I removed the card and also removed the system.conf and chan_dahdi entries. I am using ConfBridge in a PA

I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xxxxxx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to

I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -------------- Feb 17 10:59:14 WARNING[18726]:

Hi All, I have 2 tables 1] user_infos & 2] users class UserInfo has_one :user validates_presence_of :city class User belongs_to :user_info validates_presence_of :first_name i write following code in my create method. @user_info = UserInfo.new(params[:user_info]) @user=@user_info.build_user(:first_name=>'''') if @user_info.save else end now what i want is to

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each

Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI> module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec

hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using "res_timing_dahdi" or I can use "res_timing_timerfd" to get some benefit if I upgrade to 1.8? thank a lot for

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL: