Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters
From https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces: res_timing_dahdi uses timing mechanisms provided by DAHDI. This method of timing was previously the only means by which Asterisk could receive timing. It has the benefit of being efficient, and if a system is already going to use DAHDI hardware, then it makes good sense to use this timing source. If, however, there is no need for DAHDI other than as a timing source, this timing source may seem unattractive. For users who are upgrading from Asterisk 1.4 and are used to the ztdummy timing interface, res_timing_dahdi provides the interface to DAHDI via the dahdi kernel module. res_timing_timerfd uses a timing mechanism provided directly by the Linux kernel. This timing interface is only available on Linux systems using a kernel version at least 2.6.25 and a glibc version at least 2.8. This interface has the benefit of being very efficient, but at the time this is being written, it is a relatively new feature on Linux, meaning that its availability is not widespread. On 01/15/2019 09:53 AM, Thomas Peters wrote: 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 /usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1 date and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing for sure. NTPD is running, but not working for some reason. I fixed it (ownership of ntp.conf wrong) so now ntpq -pn returns a server ID. Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Stefan Viljoen Sent: Tuesday, January 15, 2019 12:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well. To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf: . . . load => res_timing_dahdi.so noload => res_timing_pthread.so noload => res_timing_timerfd.so That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines. We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had. So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything... Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it? Because the symptoms described could indicate that the clock in the VM is ju
Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters
Actually, I was wrong about that. We no longer use OVM. It's actually Citrix Xencenter 7.6 Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: Tuesday, January 15, 2019 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters >>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- 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 -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
>>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
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 /usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1 date and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing for sure. NTPD is running, but not working for some reason. I fixed it (ownership of ntp.conf wrong) so now ntpq -pn returns a server ID. Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Stefan Viljoen Sent: Tuesday, January 15, 2019 12:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well. To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf: . . . load => res_timing_dahdi.so noload => res_timing_pthread.so noload => res_timing_timerfd.so That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines. We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had. So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything... Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it? Because the symptoms described could indicate that the clock in the VM is just running too fast - or that some timing implementation detail inside Asterisk itself is running too fast. Regards Stefan -- _ -- 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 -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well. To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf: . . . load => res_timing_dahdi.so noload => res_timing_pthread.so noload => res_timing_timerfd.so That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines. We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had. So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything... Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it? Because the symptoms described could indicate that the clock in the VM is just running too fast - or that some timing implementation detail inside Asterisk itself is running too fast. Regards Stefan -- _ -- 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
