Re: [asterisk-users] Is Confbridge performance < Meetme performance
Ok, let med rephrase this: Does anyone actually uses Confbridge and what are your experiences? How about local channels in combination with Confbridge? /Johan Den 2015-09-23 kl. 13:37, skrev Johan Wilfer: Hi! I did some tests with Asterisk 11.19.0 and Confbridge. I've wanting to migrate from Meetme to Confbridge for a long time now. For two participants in a conference - one actual call and one local channel that are recording - the cpu sits at 20%. The result was the same with res_timing_pthread and res_timing_timerfd. For 15 calls the cpu sits at 200%. (res_timing_pthread) This is a Xeon E5620 @ 2.40GHz server, and with Meetme/Dahdi the load for this was hardly noticeable. Kernel: 2.6.32-openvz-042stab111.11-amd64, Debian 7 OpenVZ HN, with Debian 7 VE. And as I said - with Meetme in the same server you can barley notice the load. And with Confbridge it jumps to 20%... Any idea why Confbridge comes with such a performance hit? /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is Confbridge performance < Meetme performance
Hi! I did some tests with Asterisk 11.19.0 and Confbridge. I've wanting to migrate from Meetme to Confbridge for a long time now. For two participants in a conference - one actual call and one local channel that are recording - the cpu sits at 20%. The result was the same with res_timing_pthread and res_timing_timerfd. For 15 calls the cpu sits at 200%. (res_timing_pthread) This is a Xeon E5620 @ 2.40GHz server, and with Meetme/Dahdi the load for this was hardly noticeable. Kernel: 2.6.32-openvz-042stab111.11-amd64, Debian 7 OpenVZ HN, with Debian 7 VE. And as I said - with Meetme in the same server you can barley notice the load. And with Confbridge it jumps to 20%... Any idea why Confbridge comes with such a performance hit? /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi modules error with OpenVZ kernel on Debian 7 (2.6.32-openvz-042stab111.11-amd64)
Hi! After upgrading kernel from 2.6.32-openvz-042stab108.8-amd64 to 2.6.32-openvz-042stab111.11-amd64 I fail to load the DAHDI kernel modules. It's the package dahdi-linux-complete-2.10.2+2.10.2 and it looks like it compiles and installs just fine (a few error in the text indicates that something is wrong however) After "make ; make install ; modprobe dahdi" This is the output: libkmod: ERROR ../libkmod/libkmod.c:505 kmod_lookup_alias_from_builtin_file: could not open builtin file '/lib/modules/2.6.32-openvz-042stab111.11-amd64/modules.builtin.bin' FATAL: Module dahdi not found. When I look in /lib/modules/2.6.32-openvz-042stab111.11-amd64 the dahdi-subtree that exists in the /lib/modules/2.6.32-openvz-042stab108.8-amd64 (previous kernel) is missing. Output for make is attached below. Anyone that is a little bit less lost with compiling kernel modules than I am? Many greetings! /Johan --- uname -a Linux pi 2.6.32-openvz-042stab111.11-amd64 #1 SMP Tue Sep 1 18:27:11 MSK 2015 x86_64 GNU/Linux root@pi:/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' VERSION /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/xpp/xpp_version.h Building modules, stage 2. MODPOST 36 modules WARNING: could not find /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' root@pi:/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux# make install make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/firmware' make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' VERSION /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/xpp/xpp_version.h Building modules, stage 2. MODPOST 36 modules WARNING: could not find /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' build_tools/uninstall-modules dahdi 2.6.32-openvz-042stab111.11-amd64 make -C /usr/src/linux SUBDIRS=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/include DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes INSTALL_MOD_PATH= INSTALL_MOD_DIR=dahdi modules_install expr: syntaxfel make[1]: Entering directory `/usr/src/linux-headers-2.6.32-openvz-042stab108.8-amd64' INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dummy.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_eth.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_ethmf.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_dynamic_loc.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_jpah.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_kb1.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_mg2.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_sec.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_echocan_sec2.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_transcode.ko INSTALL /usr/src/dahdi-linux-complete-2.10.2+2.10.2/linux/drivers/dahdi/dahdi_vpmadt032_loader.ko INSTALL
Re: [asterisk-users] Asterisk 13 WebRTC Status report
Den 2015-09-15 kl. 16:52, skrev asandoval...@gmail.com: Hello Marek! I’ve been running on an issue with my Asterisk 12 configuration for using WebRTC on a LAN environment for about a month! I really need some help … My calls from the browser are done fine. I get ringing, they can be answered and never drop. The thing is that there is no audio on any side! But I don’t get any error or warning from JavaScript nor the Asterisk CLI. I’m using Asterisk 12 + jsSIP. If you could help me solving this I would be eternally greatful It’s for my grade project … These are my files: sip.conf: http://pastebin.com/kWwXpi4V http.conf: http://pastebin.com/ZwJWiiwf SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb SIP debugging for extension call (Hello-World recording): http://pastebin.com/0PxjLwBb I followed these tutorials. If you have any other useful resource, I’d be glad if you could share it: http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11 http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html Furthermore, if I want to have a local Asterisk configuration, which should be the IP address for the http.conf + DTLS certificates?? I tried with localhost but RTP packets redirect to my eth IP. Thanks in advance!! In asterisk you have "rtp set debug on" to see if you get rtp packets. On your client you can start wireshark and look if RTP packets flow in both directions. If you have RTP traffic, maybe you didn't attach the incoming media to an audio/video tag in your html. For example: html: In the event-handler for 'addstream' for the call, you have to attach the stream to #remoteView. /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVZ with asterisk 13
Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe 150 call) As long as you don't overload the server it works great. I've used OpenVZ to separate Asterisk instances from each other. For my application (mostly conferencing) I can put ~ 350 concurrent calls on a single HP Xeon server. OpenVZ is not really like KVM but more like Solaris containers or BSD jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but OpenVZ also has some cusom stuff. If you need Dahdi you will need to give the VE's access to these devices, there are articles out there that explain how this is done. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVZ with asterisk 13
Den 2015-04-07 20:47, Mitul Limbani skrev: With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. That's another issue thought and is not different on a dedicated server. With proper monitoring of the resources, you can troubleshoot. I've found voipmonitor.org to be invaluable in that regard. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and AMI in PHP -- What's current?
Den 2014-11-18 21:33, Steve Edwards skrev: I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? I use these, works very well: https://github.com/marcelog/PAGI https://github.com/marcelog/PAMI More modern, uses composer, does get updates. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover / modifying response time
Den 2014-09-04 18:05, Stephen More skrev: I was able to get a packet trace of this event Time 312.353549 - INVITE to primary 313.222303 - INVITE to primary ( suspected resend of frame ) 314.289215 - INVITE to backup 315.397120 - INVITE to backup ( suspected resend of frame ) So is primary just too slow to answer ? I am not seeing anything in the logs on primary. You can try with Wait(2) to wait two seconds before you do Answer(). This will delay the response to the INVITE with two seconds. You don't want to wait to long thought, so maybe you can test your way to something that works for you. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(dst) not set in AEL macro
2014-07-11 15:38, Rafael dos Santos Saraiva skrev:
Hi
I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field CDR(dst), showing only ~~s~~.
I tried various configurations, but without solutions.
This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}
I don't know if this is maybe related to this:
https://issues.asterisk.org/jira/browse/ASTERISK-20441
If it is this is a bug in the AEL compiler I think.
--
Johan Wilfer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...
2014-04-15 10:37, Lee, John (Sydney) skrev: Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release: 1) Asterisk 1.4.21.2 2) Libpri-1.4.4 3) Zaptel-1.4.11 I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6, 1.8, 10.x, 11.x, 12.x, 13.x Can someone please give me some advice as to what release I should upgrade? Or should I just stick to 1.4.x and just upgrade DAHDI? Thanks. Regards, John Lee The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 1.4, and 1.6-series have no support anymore. 1.8 is an LTS and have support currently, but this is also true for 11 and asterisk 11 will be supported longer. You have the full list here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions I would go for Asterisk 11 in your case. You will have to think of it more like a migration than an upgrade thought, as a lot has happened since asterisk 1.4. On a side-note, I still run some old installations with a current Dahdi + Asterisk 1.4.44 and they work great together. There is the non-support catch however. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Webrtc and adventures with Asterisk 11
Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-0005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101 -- Asterisk sends SIP/2.0 488 Not acceptable here Chrome: I've tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video with confbridge to a handful of participants seems to use quite a bit of cpu thought. Screen-share: This works, but Confbridge is not very happy about a channel with video (vp8) and not audio and is printing this 80 times a second: WARNING[8919][C-] channel.c: Unable to find a codec translation path from (vp8) to (slin) WARNING[8919][C-] chan_sip.c: Asked to transmit frame type slin, while native formats is (vp8) read/write = unknown/unknown WARNING[8919][C-] channel.c: Don't know any of (vp8) formats How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip? How do you use / plan to implement webrtc in your environment? Any feedback is welcome. Thanks! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAGI
2014-04-10 15:02, Gopalakrishnan N skrev: Thanks Johan. Are you using this application for any credit card processing? Nope, mostly ChannelRedirects and injecting sounds in conferences / controlling via webgui. But the libraries are very good and seemed very complete to me. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 under VMware?
Hi! Anyone that have tried using Asterisk 11 with SIP + Confbridge as a VMware virtual machine? Any issues to be aware of? Of course the hardware node needs to to be powerful enough - but say you have just one virtual machine on the node - will the performance be drastically less than running asterisk on the metal? Or can I expect roughly the same performance? Thanks! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAGI
2014-04-03 18:58, Gopalakrishnan N skrev: Hi, Anybody using PAGI scripts, http://marcelog.github.io/articles/pagi_tutorial_create_voip_telephony_application_for_asterisk_with_agi_and_php.html Would like to know the feasibility to build a IVR solutions. Regards I use PAMI, and it works great. PAGI seems to be a sister-project for AGI. https://github.com/marcelog/PAMI https://github.com/marcelog/PAGI -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 under VMware?
2014-04-04 19:30, Carlos Chavez skrev: I have found Asterisk using only SIP is very responsive on virtual machines. We have used VMs for call center applications and for complex IVR solutions without problems. Obviously there is overhead running a VM so you can never expect a VM to perform as well as bare metal. Running a single VM on a server is a complete waste of resources, might as well run natively. Sounds very good. Do you have this experience with WMware in particular or with virtualization in general? I won't run a single WM, it was just an example. My question was more about if I could expect roughly the same performance, or if it is drastically different with virtual machines on VMware. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 under VMware?
2014-04-04 19:30, Carlos Chavez skrev: I have found Asterisk using only SIP is very responsive on virtual machines. We have used VMs for call center applications and for complex IVR solutions without problems. Obviously there is overhead running a VM so you can never expect a VM to perform as well as bare metal. Running a single VM on a server is a complete waste of resources, might as well run natively. Thanks for the feedback! Do you have this experience with WMware in particular or with virtualization in general? I won't run a single WM, it was just an example. My question was more about if I could expect roughly the same performance, or if it is drastically different with virtual machines on VMware. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 under VMware?
2014-04-04 21:35, Kevin Larsen skrev: From: Johan Wilfer li...@jttech.se Sounds very good. Do you have this experience with WMware in particular or with virtualization in general? We run our Asterisk 11 instance in VMWare as well. They share the hardware with multiple other boxes. We do give Asterisk priority over most other virtual machines. We either have SIP providers or use boxes like Digium's G100 series to convert our T1 lines to SIP. Our experience has been good and we have no problems loading Asterisk up on virtual machines on each site. Thanks Kevin, that's great. Nice to hear that asterisk is more virtualization friendly with recent versions. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confbridge options
Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading User Profile Configuration Options the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty conference. On by default. -- But - disabling it (announce_only_user=no) doesn't take effect. And looking at the source asterisk-11.8.1/apps/app_confbridge.c I can't even find this option. Any clues? Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an oversight? -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge options
2014-04-04 22:01, Johan Wilfer skrev: Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading User Profile Configuration Options the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty conference. On by default. -- But - disabling it (announce_only_user=no) doesn't take effect. And looking at the source asterisk-11.8.1/apps/app_confbridge.c I can't even find this option. Any clues? Looked at the wrong file for config parsing, sorry for the noise. But the option is not respected thought. Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an oversight? Reading the source I get the impression that the intended behavior is: 1. Read sound_only_person if not flags quiet or announce_only_user is set. (apps/app_confbridge.c:1099) 2. No join sounds if flag quiet is set. (apps/app_confbridge.c:1714) But this is not what happens with 11.8.1, this is the bridge/user: [conference_bridge] type=bridge [conference_user] type=user admin=no marked=no startmuted=no announce_only_user=no quiet=yes With this I get join sounds played, and only_user is announced as well.. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge options
2014-04-04 23:33, Johan Wilfer skrev: Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an oversight? Reading the source I get the impression that the intended behavior is: 1. Read sound_only_person if not flags quiet or announce_only_user is set. (apps/app_confbridge.c:1099) 2. No join sounds if flag quiet is set. (apps/app_confbridge.c:1714) But this is not what happens with 11.8.1, this is the bridge/user: [conference_bridge] type=bridge [conference_user] type=user admin=no marked=no startmuted=no announce_only_user=no quiet=yes With this I get join sounds played, and only_user is announced as well.. I should have gone to sleep I think, my brain doesn't work. I think I get it now however. Short version: With dynamic user profiles, CONFBRIDGE(user,template)=user_profile) must be used. The profile supplied in ConfBridge is ignored and I missed that. The end. Long version: This works: ConfBridge(1,conference_bridge,conference_user); dev02*CLI confbridge list 1 ChannelUser Profile Bridge Profile Menu CallerID Muted == = SIP/jttech_sip2-0004 conference_user conference_bridge No But this does not: Set(CONFBRIDGE(user,startmuted)=no); ConfBridge(1,conference_bridge,conference_user); dev02*CLI confbridge list 1 ChannelUser Profile Bridge Profile Menu CallerID Muted == = SIP/jttech_sip2-0005conference_bridge No I expected Confbridge to use the supplied user_profile as a template and overlay the specific settings I set with CONFBRIDGE(user) on top of that profile. But instead it seems like the default profile is used. This however works (I should have read the docs more carefully): Set(CONFBRIDGE(user,template)=conference_user); Set(CONFBRIDGE(user,startmuted)=no); ConfBridge(1,conference_bridge); dev02*CLI confbridge list 1 ChannelUser Profile Bridge Profile Menu CallerID Muted == = SIP/jttech_sip2-0007 conference_user conference_bridge Yes With dynamic user profiles, CONFBRIDGE(user,template)=user_profile) must be used. The profile supplied in ConfBridge is ignored if one is present. I didn't expect that... -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] process asterisk stop
2014-04-03 05:41, Павел Чашков skrev: Who can help? I have Asterisk 1.8.3 server on Ubuntu 10.04. Asterisk periodically falls here with this error: [1227952.625701] asterisk [30237]: segfault at 18 ip 7ff3504579bc sp 7ff34ddc3ff0 error 4 in libc-2.11.1.so [7ff3503e +17 a000] -- Pavel Chashkov www www.ngs.ru http://ngs.ru e-mail p.chash...@office.ngs.ru mailto:p.chash...@office.ngs.ru телефон (383) 212 52 52 (747) телефон +7 923 229 97 70 icq 229 794 267 Try upgrading asterisk to 1.8.26.1 that is the latest release in the 1.8-series. You are hitting a bug somewhere and odds are that it is already resolved in a later version. http://www.asterisk.org/downloads/asterisk/all-asterisk-versions -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need more meetme users -- hitting some limit
2014-03-21 18:54, Steve Totaro skrev: I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of *DAHDI linux limits the number of open pseudo channels to 512 for this reason*. [1] Thanks, Steve T I would check /proc/interrupts also. On some distros irq's are not balanced by default and are all hitting the same core. On Debian I had to install the irqbalance package and the load was spread across the cores. Dahdi is still a single thread thought. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
2013-11-13 11:55, Jonas Kellens skrev: On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Here is some suggestions to get started: http://www.enterprisenetworkingplanet.com/unified_communications/troubleshooting-common-sip-problems-with-wireshark.html Maybe one of your connections get congested? For example, if the two endpoints is your phone and the upstreams teleco. If the side from the teleco are bad and not the phone you need to take a closer look at the switches and routers on the way to the teleco. For example you can run tcpdump on your gateway to your ISP. If you see the problem here as well it may be your link or a upstreams problem. If you don't see it here it is somewhere in between.. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS E1 signalling
2013-10-17 17:48, Russ Meyerriecks skrev: On Thu, Oct 17, 2013 at 7:53 AM, Johan Wilfer li...@jttech.se mailto:li...@jttech.se wrote: 1. In asterisk can I get the channel-number of the call so I can have different logic for the different channels? Sure, I guess I would just create different incoming contexts for your various channels in chan_dahdi.conf. Or you could write some dialplan foo. 2. How do I handle answer / hangup with CAS? Will DAHDI keep this channels up, or should I query the state of the channels (how?) and bring them up myself if they are down (Dial?) DAHDI will interpret the CAS signalling and pass those up to userspace as hookstates. All of this should be seamless to you as an Asterisk user. The usual channel Answer() and Hangup() is all that's needed, just like on any other type of channel. Have fun I will :-) Thanks! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
2013-10-03 09:52, Johan Wilfer skrev: 2013-10-02 17:12, Shaun Ruffell skrev: On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote: If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. Hmm...this is the first report I've heard of dahdi_dummy being more performant than the core timer. I wonder if this has something to do with the fact that you're running under 2.6.32-5-openvz-amd64 which might be doing more work in the system timer (which is where the standard core timer work is processed). If you update to the latest 2.6.32-openvz kernel do you still have the audio problems in conferneces? I don't think I dare to make such a big change to production servers as dahdi_dummy works fine (for the users - they are the one that counts). What I have noticed from dahdi_dummy is that cpu0 is nearly 95% at ~250 channels and that got me worried (perfect quality in the meetme's thought). When you explictly load the dahdi_dummy module, your results can change in a couple of ways. 1) dahdi_dummy tries to always schedule the system timer to fire at 1ms intervals (which it only will if the system is configured for CONFIG_HZ=1000). 2) If on a newer kernel, dahdi dummy will use kernel high resolution timers to increase the precision of the timer. However this shouldn't be necessary since the jitter in the normal kernel timer should be small compared to all the other jitter in a voip system. After som more tests I noticed that the core timers work good with low load. But at ~200 concurrent calls *some* of the calls sounds bad. Very strange.. Switching back to dahdi_dummy solved this. I'll test this more with a newer kernel, but I'm unable to do that right now. Also irqbalance solved the cpu issue (see below), so that wasn't DAHDI. When I compare dahdi core timers and dahdi_dummy side-by-side I notice that dahdi_dummy spends a litte more time doing sys cpu ~10-15% for 200 calls. And with this (old) kernel it seems more tolerant to load. On a sidenote, when I investigated the 95% cpu on the first core I did notice that all irq are hitting cpu0 (/proc/interrupts). I did some reading and installed irqbalance and now interrupts are spread evenly among the cores. Can this cause issues for the core timers? After running test for some time my conclusion is that irqbalance on debian prevents the cpu from spiking on a busy system. So this solves the cpu issue. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAS E1 signalling
Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 EM. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 EM - MUX - WAN - MUX - Digium TE220, Asterisk I've found some documensts describing the Dahdi configuration for CAS E1, but I have some unanswerd questions: 1. In asterisk can I get the channel-number of the call so I can have different logic for the different channels? 2. How do I handle answer / hangup with CAS? Will DAHDI keep this channels up, or should I query the state of the channels (how?) and bring them up myself if they are down (Dial?) 3. Other suggestions? CAS is unknown territory for me so I appreciate all the pointers you have. Thank you! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Capture Media IP in CDR
2013-10-14 20:05, CDR skrev: Right now,there is no way know to capture the Media IP. I've seen serval suggestions for you on the list. I suggest you go back and read them again. Gareth Blades even handed you a solution to get sip-traces with all the signaling. That's a good solution. (I think I will use that myself for some cases - like integrated debuging in web-gui, thanks!) The sip callid doesn't change and you can filter the pcap for this. Otherwise you have more full-fledged solutions like voipmonitor.org, or kamilio with Homer. More complex but also more capable. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
2013-10-02 17:12, Shaun Ruffell skrev: On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote: If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. Hmm...this is the first report I've heard of dahdi_dummy being more performant than the core timer. I wonder if this has something to do with the fact that you're running under 2.6.32-5-openvz-amd64 which might be doing more work in the system timer (which is where the standard core timer work is processed). If you update to the latest 2.6.32-openvz kernel do you still have the audio problems in conferneces? Okay, I guess it is just me then, that's a good thing :-) As Debian have dropped support for openvz in the current release and squeeze will not recive support after 2014-05 I will need to update these machines anyway soon - that means a new kernel, and it seems every project I found uses the redhat-kernel. I don't think I dare to make such a big change to production servers as dahdi_dummy works fine (for the users - they are the one that counts). What I have noticed from dahdi_dummy is that cpu0 is nearly 95% at ~250 channels and that got me worried (perfect quality in the meetme's thought). - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? This is because when using the core timer, the timer is only scheduled to fire ever 4ms. The differences in each *individual* measurement you see is due to timer jitter + the increased interval leaking more of the slight jitter up to userspace. However, this isn't typically a problem when mixing audio in 20ms chunks by default as is typically done when you're using meetme conferences. The number that is generally more interesting is the Cummulative Accuracy which shows over the entire dahdi_test how close DAHDI was to processing the expected amount of audio. On another system with 10 ms timer ticks the jitter is increased, but even this system does not have any problems mixing audio in meetme conferences: Best: 99.608% -- Worst: 99.418% -- Average: 99.531611% Cummulative Accuracy (not per pass): 99.995 When you explictly load the dahdi_dummy module, your results can change in a couple of ways. 1) dahdi_dummy tries to always schedule the system timer to fire at 1ms intervals (which it only will if the system is configured for CONFIG_HZ=1000). 2) If on a newer kernel, dahdi dummy will use kernel high resolution timers to increase the precision of the timer. However this shouldn't be necessary since the jitter in the normal kernel timer should be small compared to all the other jitter in a voip system. Thanks for this detailed explination of the inner workings of dahdi! There are statements (mostly very old) that say that the results should not drop below 99.9 or you will have quality problems. My observation of quality problems in connection with results at 99.6 seemed to confirm this, but if I understand you correctly dahdi should tolerate this. - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? I personally do not think so. The most efficient way should just be to allow the normal kernel timers to also provide timing to your asterisk system without the overhead of processing another interrupt. So I guess I would be interested to hear if a kernel update is all you need to have good results with the core timer. Seems very likley. I'll start running tests at the replacement-system. But now I know how to troubleshoot this problem. On a sidenote, when I investigated the 95% cpu on the first core I did notice that all irq are hitting cpu0 (/proc/interrupts). I did some reading and installed irqbalance and now interrupts are spread evenly among the cores. Can this cause issues for the core timers? -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi_dummy is more accurate than core timer?
Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Below is some stats (trimmed, but you get the idea). Thanks! /Johan ** With Dahdi 2.7.0.1, and core timers: 99.998% 99.611% 99.615% 99.997% 99.993% 99.997% 99.996% 99.608% 99.999% 99.612% 99.607% 99.613% 99.999% 99.998% 99.994% 99.609% --- Results after 177 passes --- Best: 100.000% -- Worst: 99.604% -- Average: 99.901099% Cummulative Accuracy (not per pass): 99.998 ** With Dahdi 2.7.0.1, and dahdi_dummy loaded: 99.993% 99.998% 99.998% 99.993% 99.996% 99.998% 99.996% 99.998% 99.998% 99.997% 99.999% 99.998% 99.996% 99.998% 99.999% 99.997% --- Results after 177 passes --- Best: 100.000% -- Worst: 99.993% -- Average: 99.997738% Cummulative Accuracy (not per pass): 99.998 ** With Dahdi 2.7.0.1, and Wildcard TE220 providing timing 99.981% 99.983% 99.983% 99.982% 99.983% 99.982% 99.984% 99.981% 99.982% 99.983% 99.984% 99.981% 99.980% 99.984% 99.983% 99.983% --- Results after 177 passes --- Best: 99.996% -- Worst: 99.974% -- Average: 99.982104% Cummulative Accuracy (not per pass): 99.982 Kernel: 2.6.32-5-openvz-amd64 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
2013-10-02 13:55, Gareth Blades skrev: On 02/10/13 12:17, Johan Wilfer wrote: Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Its a little different when you are using meetme as its an application built into dahdi itself and not a native asterisk application. It will therefore always use dahdi for its timing. If dahdi doesnt have a hardware interface (sangoma sell a usb based timing source if you want a hardware source) then it will use a software timing source of some form. I dont know what method it uses. Yes, this is for a legacy application that are using Meetme. Maybe I was unclear above but with core timer I meant just modprobe dahdi. dahdi_dummy = modprobe dahdi_dummy and the hw card used wct4xxp. More information about timing sources can be found at :- https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces The wiki states: As of DAHDI Linux 2.3.0 the dahdi_dummy module has been removed and its functionality moved into the main dahdi kernel module. As long as the dahdi module is loaded, it will provide timing to Asterisk either through installed telephony hardware or utilizing the kernel timing facilities when separate hardware is not available. But when I test just dahdi (core timer, no dahdi_dummy) I get distortion and bad quality in the Meetme-conferences. This does not happen with dahdi_dummy. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bind to ipv4 ipv6
http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB Partial Replication?
2013-09-20 21:48, Doug Lytle skrev: Any takers? astdb is based off of version 1 BerkeleyDB. Googling shows: http://www.voip-info.org/wiki/view/Asterisk+database It has a section on basic replication. If you need to do this, you probably would be better off with a real database. Look at asterisk's func_odbc and maybe also database replication. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI wct4xxp high system CPU on idle?
2013-08-14 19:48, Tony Mountifield skrev: I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't relevant to the question). So my first question would be: is this high CPU usage normal with current cards and DAHDI? It's curious that 12.5% is 1/8 of 100% and /proc/cpuinfo reports 8 CPUs, but I don't know whether that is just coincidence. The CPU is a X3450 with four cores and HT enabled. I had strange issues with DAHDI and HT enabled. This was with Debian 6 about a year ago with a Digium 4 and 8-port PCI-E E1 card, wct4xxp driver. The symptom was choppy sound for all calls. The problem was resolved by disabling hyper-threading. From that moment on I always disable HT, to avoid problems. I havn't seen your specific 12.5% cpu problem thought. Maybe worth a shoot anyway? -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme and maxusers option
On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote: 2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. 2013-07-22 16:59, Thiago Coutinho skrev: Hi Johan. But the option maxusers should work too, right? I guess so, but I have not used it myself. It's not very hard to build you own dialplan with func_odbc and custom tables. This way you could use Meetme, Confbridge, or something else to do the mixing. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme and maxusers option
2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip
2013-07-01 15:04, Daniel-Constantin Mierla skrev: Hello, On 6/28/13 4:29 PM, Johan Wilfer wrote: Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the external. This is the setup: Teleco - Kamailio - Asterisk SIP -- 1.2.3.4 10.0.0.1 -- 10.0.0.2 externip=1.2.3.5 localnet=10.0.0.0/255.255.255.0 RTP 1.2.3.5 (NAT:ed to 10.0.0.2) On an incomming call from the teleco - to kamailio (public addr) - to asterisk in the private net. Asterisk responds with the following SDP: v=0 o=root 1889 1889 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 23344 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Asterisk seems to think that because the proxy is on the localnet, the media is too, so it doesn't use the externip as the RTP-ip. This is a incomming call and the RTP ip of the other leg is another public address. So the RTP-ip should the public address (externip). If I connect to the teleco directly from the pbx (bypassing kamailio) Asterisk correctly uses the externip as the rtp-ip in the SDP. I know this is an old and unsupported version of Asterisk, but any input on the topic is welcome. If this is supported in later versions we can maybe work around until we migrate later. what I did when I had similar scenario was to let asterisk completely behind NAT, using only the local IP. I used rtpproxy running on the same host as kamailio to bridge the rtp between external and internal networks. Cheers, Daniel I think that you are right that this should be done with Kamailio. Maybe the nathelper-module in Kamilio would do the trick in modifying the SDP/Contact to the NAT:ed address instead of using rtpproxy. Thanks for the feedback! /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 crashing on x64 when transferring a call
2013-06-28 23:33, Steve Edwards skrev: Yes, I know, but I'm a 1.2 Luddite. 1.2 on 32 bits has been stable and rock solid for almost 10 years, but the client wanted to try it on a 64 bit version of CentOS 5.9. It all seems to be working OK, but when the customer service agents transfer a call they got from a queue, Asterisk crashes. Has anybody else had issues running 1.2 (or similarly ancient versions) on 64 bit OSes? Can anybody remember back that far? I've used the latest 1.4 on amd64 (Debian 6) since the beginning of 2012. Before that I was running Debian 5 (32 bit). Rock-solid and no crashes at all, but no transfers either. Maybe you can use 32-bit with PAE? This way you can use up to 64G memory in the server, but maximum 4G per process. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip
Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the external. This is the setup: Teleco - Kamailio - Asterisk SIP -- 1.2.3.4 10.0.0.1 -- 10.0.0.2 externip=1.2.3.5 localnet=10.0.0.0/255.255.255.0 RTP 1.2.3.5 (NAT:ed to 10.0.0.2) On an incomming call from the teleco - to kamailio (public addr) - to asterisk in the private net. Asterisk responds with the following SDP: v=0 o=root 1889 1889 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 23344 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Asterisk seems to think that because the proxy is on the localnet, the media is too, so it doesn't use the externip as the RTP-ip. This is a incomming call and the RTP ip of the other leg is another public address. So the RTP-ip should the public address (externip). If I connect to the teleco directly from the pbx (bypassing kamailio) Asterisk correctly uses the externip as the rtp-ip in the SDP. I know this is an old and unsupported version of Asterisk, but any input on the topic is welcome. If this is supported in later versions we can maybe work around until we migrate later. Thanks! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'n Dahdi on Sun Solaris
2013-06-12 11:42, Chandrakant Solanki skrev: Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. If it's a new application you are building - Why not test asterisk 11 + confbridge? This way you won't need DAHDI. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic
2013-06-06 22:21, motty cruz skrev: Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104@sipphones:1] MeetMe(SIP/101-0813, 104) in new stack == Parsing '/etc/asterisk/meetme.conf': == Found [Jun 6 13:17:30] WARNING[11457]: app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo device Meetme is depending on DAHDI, and have always been. Asterisk would not even compile Meetme if DAHDI header files isn't present. So you must have installed DAHDI at some point? Try run dahdi_test (after installing), you should get output like 99.9% 99.8% ... and so on. Then you know the timing works (dahdi psuedo) -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem to install asterisk on vps digitalocean
2013/6/4 James Cloos cl...@jhcloos.com mailto:cl...@jhcloos.com t == troxlinux xserverli...@gmail.com mailto:xserverli...@gmail.com writes: t I try to install asterisk on vps server , but fails when I want to t install dahdi There is no hardware for dahdi to use; you shouldn't need to install it. 2013-06-04 22:28, troxlinux skrev: if it is true I have not any hardware but I need help to solve it and I think it could serve other future Depending on the type of VPS this can work or not. My general experience of normal stock-VPS:es are that they are often oversold. And it will create a lot of issues for you with voip. On OpenVZ you can't load kernel-modules in the VE anyway, so it's no idea to compile Dahdi. But if you control the HN, you can install it there and give the VE access to it. In that case you should compile it in the VE as well. The same applies to LXC. On XEN/KVM I think it will work, but be sure to test the timing with dahdi_test - you don't want the results to drop below 99.9. Try create some i/o (for example compile asterisk) and run the test in another window. Dahdi in VE's is a pain, but can work if you control the HN. You should do development and testing on a dedicated server. After that, try move to a VPS and try to fix the issues. The other way around is very hard. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing on an OpenVZ instance
2013-05-06 20:48, James Wystead skrev: Hello All; I'm attempting to build the dahdi on an OpenVZ instance: Linux serverx 2.6.18-274.7.1.el5.028stab095.1 #1 SMP Mon Oct 24 20:49:24 MSD 2011 x86_64 x86_64 x86_64 GNU/Linux Now, the kernel says that I have the proper one installed, as you can see from above. However, when I run the make all, this is what I see: You do not appear to have the sources for the 2.6.18-274.7.1.el5.028stab095.1 kernel installed So, my question is this - what is the best way to fix this? Feel free to ask anything you want as I really want to get this working. You probably need the header-files for the kernel. in debian it is: apt-get install linux-headers-`uname -a` in centos in general: yum install kernel-devel I don't have very much experience with Centos but I thinks this page have some suggestions worth trying out (ovzkernel-devel package): http://stackoverflow.com/questions/11592010/compile-dahdi-on-openvz-vps-kernel-issue Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debug strategy for one-way audio calls
2013-05-02 13:19, Marie Fischer skrev: Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Apart from logging all traffic 24/7 via tcpdump (not really convenient), can you give me some ideas how to debug this kind of issue? Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters. Voipmonitor.org is great for debugging voip. You can either use only the sniffer (opensource) and use mysql + the pcap files or you can also buy the commercial webgui. Either way, it's a great product. /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to simulate public IPs for lab testing
Please, excuse me but I'm not sure I got your suggestion and I'm realizing I didn't correctly describe my lab set up. At the moment, the router between both servers provides Internet access to server1. That means it has one WAN interface eth0 which is on server2 side and one eth1 LAN interface which is on server1 side. Currently, this router do NAT translation for server1. Having clarified my setup, I guess your advice is to : 1. add address 1.2.3.4 to router's eth0 2. add address 4.3.2.1 to server2 interface 3. configure router to route trafic to 4.3.2.1 using server2 private address (such as ip route add 4.3.2.1/24 http://4.3.2.1/24 via 192.168.1.25) 4. configure server to route trafic to 1.2.3.4 using router private address Is this correct ? Sorry for the delay, but yes, that's how I do it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to simulate public IPs for lab testing
2013-04-08 16:36, Olivier skrev: Hello, Many times, I need to test in a lab Asterisk servers before sending them to customer locations. I'm currently having trouble to test SIP trunks without touching SIP configuration. So, how should I change my testing lab so that I could now test SIP trunks without modifying Asterisk server under test ? A typical set up is: Asterisk server1 under test ---SIP Router - SIP Lab's Asterisk server2 All machines (server1, router and server2) have Internet access. Router and server2 have a private address. Ideally, router should get customer's public adress (eg 1.2.3.4), server2 should also get my ITSP public address (eg 4.3.2.1) and both machines should route trafic to each other without leaving my LAN and using Internet access. What would you suggest ? I often configure a router to do NAT in these cases. You can do NAT even with a public net on the inside. Configure the temporary router with the IP of the customers router for the inside, and make it a dhcp client (or whatever you use) in your LAN for the outside interface. You can make a route in your router to your ITSP-gw like this : route add -host 4.3.2.1 dev eth0 This means all traffic to 4.3.2.1 will go to dev eth0 on your router. (The device in your network with the ip 4.3.2.1 also needs to have a route back to your router for replies.) Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] crossed channels
2013-02-19 17:10, Juan Carlos Agudelo skrev: I don't have analog channels, this happens with SIP Trunk... Juan.. I've seen this with one of our sip-trunks. Our provider used opensips for that platform I think. If had the same account registered in two asterisk-servers and they answered the call at the same time, audio was from both asterisk-servers and the phone from pstn, like a 3-way conference. I also watched a presentation about rtp ports, mentioned on this list some time ago, explaining quite a bit on rtp ports and security: http://lists.digium.com/pipermail/asterisk-users/2013-January/277342.html http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html So my guess is that if you get two devices using the same port, or one device that don't stop sending, you will hear that injected in your call. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for SIP to ISDN PRI E1 gateway to use with
2013-02-15 22:32, Mc GRATH Ricardo skrev: Hi John How about Digium G100 G200 voip gateways? You can try with a test drive at Digium web site http://www1.digium.com/en/products/voip-gateways Best regards. Thank you! Did you use them yourself? I've looked at the Digium gateways, and they look very promising. I've used the TE420/TE820-cards previously, and they just keep working.. :-) -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Split SIP and RTP to different IP addr
2013-02-15 11:26, Mikhail Lischuk skrev: Greetings! I have an Asterisk 1.4 box and due to hardware limitations I cannot upgrade atm. So, as long as I understood from different posts, SIP-TLS is not available for 1.4 Then I set up VPN and route all inter-Asterisk traffic into VPN. But for some reason, with all the RTP inside the VPN I start getting packet losses up to 30%. Maybe CPU is too weak, that is yet to be discovered. What I want to ask is - how can I split SIP and RTP traffic? Say, SIP goes via VPN, but after the call is initiated, servers reinvite each other with real IPs. Is that possible at all? Searching on Internet didn't give me a clue. You probably wants a SIP Proxy (like Kamailio). This way you can have SIP signalling over VPN or use TLS, and kamailio can talk with asterisk over udp. RTP always flows directly between asterisk and your provider, and sip will use the proxy: SIP: Provider - (vpn/tls) - Kamailio - (udp) - Asterisk RTP: Provider - -- - Asterisk Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendations for SIP to ISDN PRI E1 gateway to use with Asterisk?
Hi, Anyone who has experience with ISDN PRI voice gateways? My customer wants to connect some E1 equipment to a gateway that converts from ISDN PRI E1 to SIP/RTP. The data will be transmitted over a WAN, and into an Asterisk-1.8 server. It's one E1 on each site at 8 sites, and they are asking for our recommendations on gateways. Your advice on this topic is very appreciated! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a need to secure RTP ports?
2013-01-23 18:20, Sebastian Arcus skrev: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). You could try iptables with ip_conntrack_sip ip_nat_sip. If they are loaded and you accept calls from your sip provider on port 5060 iptables inspects the sip/sdp and traffic from the endpoints are considered RELATED. I've some research/testing to do myself on this topic (it's on my always growing todo-list of doom.. :-) Maybe you should check it out? -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special conference room
2013-01-16 22:10, Yves A. skrev: barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... does _almost_ all features.. unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the mute switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones am I wrong? nevertheless... any more hints for my original feature-request? thank you all, yves I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. You can do all this with Meetme, ChanSpy and ChannelRedirect. You could also use the AMI variants of the above commands. Maybe you could use Confbridge that is intended to replace Meetme. So the simple answer is yes, this can be done. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php programming for working with asterisk
2013-01-14 09:43, Muhammad skrev: /thanks to replay Sammy! But excatly I don't know how can do it! connecting to DB via dialplan. Here is a good example explaining this: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/getting_funky.html It's written for asterisk 1.4, but I think that will get you started. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the maximum number of meetme's allowed?
2012-12-24 16:13, Deepesh D skrev: Hello, What is the maximum number of meetme's allowed by asterisk. On my server with an 8 GB memory, I start getting the following error after 150-160 meetme's are created WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo channel: Cannot allocate memory At this time the server still has about 6 GB of free memory. I even tried this on a server with higher memory, it gives the same result. I am using asterisk 1.4.44. You have probably run out of file descriptors. Try ulimit -n 8192 before starting asterisk (or in the safe_asterisk-script or the init.d-script). I think this is per default 1024 on debian, and if you use sip + meetme you will hit the limit with about 150 concurrent calls. -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Web: www.jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing complex setups with Asterisk
2012-11-08 00:26, Jeff LaCoursiere skrev: On 11/07/2012 05:20 PM, Jeff LaCoursiere wrote: On 11/07/2012 02:16 PM, Johan Wilfer wrote: 2012-11-07 20:49, Jeff LaCoursiere skrev: Just to chime in, if you REALLY want multi-tenant, it is super easy and surprisingly efficient to use kernel level virtualization to run multiple instances of asterisk (and even FreePBX). We use LXC to do this. The host runs an instance that has the dahdi hardware, drivers, and upstream connections. The clients have SIP connections to the host for all inbound/outbound, so you have a central place to collect/process CDR records for billing. Getting your phones to connect to each instance is an exercise for the network admin ;) Any quirks / observations you have running LXC? We run OpenVZ now with the same setup and it works very well. But as Debian will not support OpenVZ in the next version we are looking for alternate solutions.. Do you run Dahdi run Dahdi for timing / meetme on both the host (HN) and the clients (VE)? Distribution? Any other pitfalls or recommendations with LXC? Since moving to Ubuntu 12.04 server, LXC mgmt has been much simpler and stable. Had some troubles with Ubuntu 10, though that was our proof-of-concept, and mainly just with getting a template finalized. Shutting down a container, back then, was a scary and often fatal thing to do. WIth 12.04 I have had zero LXC related issues in roughly six months. Have a few dozen companies running on the platform and getting ready to white label the infrastructure to several resellers. In our lab we have managed to get 200 instances, with FreePBX, running simultaneously (though idle) on one host. Each (optimized!) container seems to eat about 75M of RAM. Our latest tweak is to make all of the containers internally addressed on an OpenVPN-only accessible virtual LAN, and are only distributing telephony hardware that can connect to the platform natively (still on a search for the right ATA, though getting by with DD-WRT router in front of Cisco ATA). Cheers, j Realized I didn't answer your questions. Yes, the host runs Dahdi for timing, entirely for Meetme in the containers (host is just for transport), and the driver is exposed to the containers in the LXC conf file. LXC does NOT have the same level of operational controls over the containers, yet, as OpenVZ (like limiting resources). That hasn't really been an issue for us so far. Cheers, j That's very cool. I've also looked at Ubuntu server for their long support-time. I definitely have to test this setup. Thanks for the insight! -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Web: www.jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing complex setups with Asterisk
2012-11-07 20:49, Jeff LaCoursiere skrev: Just to chime in, if you REALLY want multi-tenant, it is super easy and surprisingly efficient to use kernel level virtualization to run multiple instances of asterisk (and even FreePBX). We use LXC to do this. The host runs an instance that has the dahdi hardware, drivers, and upstream connections. The clients have SIP connections to the host for all inbound/outbound, so you have a central place to collect/process CDR records for billing. Getting your phones to connect to each instance is an exercise for the network admin ;) Any quirks / observations you have running LXC? We run OpenVZ now with the same setup and it works very well. But as Debian will not support OpenVZ in the next version we are looking for alternate solutions.. Do you run Dahdi run Dahdi for timing / meetme on both the host (HN) and the clients (VE)? Distribution? Any other pitfalls or recommendations with LXC? -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Web: www.jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi dummy
2012-10-23 16:28, Jerry Geis skrev: I need to use the dahdi dummy driver. Its not being compiled at this time. When I go into tools subdirectory under dahdi-linux-complete-2.4.1 How can I get the dahdi_dummy.c driver compiled? I run dahdi-linux-complete-2.6.1+2.6.1.tar.gz on Debian, and in 2.6.1 dadhi_dummy is compiled with the rest of the modules. Maybe you should try updating Dahdi? to load the driver you just do: modprobe dahdi_dummy and to verify it is loaded: lsmod | grep dahdi I've run some tests with Meetme and dahdi_dummy seems to consume less system resources (cpu) than the build in timing in dahdi that is used if you don't load dahdi_dummy. Good luck! -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Phone: +46 31 3809100 Web: www.jttech.se | Mail: jo...@jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reuse h extension?
2012-09-29 11:32, Stefan at WPF skrev: I have 2 contexts, however both have the same h extension. Currently I am doing copypaste for the h extension - is there a better way? Can I somehow reference a h extension, so I have to create/modify it only once? Thanks for any hint! Have you tried goto? I have some extensions that are related and I use goto to the main context from the others. Goto(context,h,1) -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI Permissions, all means different things?
2012-09-07 16:13, David M. Lee skrev: On Sep 7, 2012, at 1:49 AM, Johan Wilfer wrote: Hi! I'm trying to limit the permissions for a AMI-account. But I'm a little bit confused by the permissions. The commands I use are (output from manager show commands, btw: privilege col seems cropped?): Yes, sadly it is. Action PrivilegeSynopsis Redirect call,all Redirect (transfer) a call. Originateoriginate,allOriginate a call. Getvar call,reporting, Gets a channel variable. If I put this in my manager.conf: [pbx_ami] secret = *** deny=0.0.0.0/0.0.0.0 permit = x.x.x.x/255.255.255.255 write=originate,call read= I get this (manager show user pbx_ami): username: pbx_ami secret: Set acl: yes read perm: none write perm: call,originate,all displayconnects: yes Where does the all permission come from? Probably just a bug in the 'manager show user' command. The user doesn't have all the permissions, so 'all' shouldn't show up in the list. If it's not already in the issue tracker, please file a bug[1]. [1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines However, If I change the row in manager.conf to write=originate,call,all the output is: username: pbx_ami secret: Set acl: yes read perm: none write perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,all displayconnects: yes Can someone explain this please? This is at least looks correct. The 'all' permission is a superset of, well, all the permissions. The 'write=all' line in manager.conf assigns all of these permissions to the user. Thanks! -- Johan Wilfer Thank you David for the feedback. I reported the following bugs: https://issues.asterisk.org/jira/browse/ASTERISK-20397 (all bug) https://issues.asterisk.org/jira/browse/ASTERISK-20396 (cropped col) -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)
2012-09-10 18:13, Tony Mountifield skrev: I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one SIP phone extension to another, and the dialplan sets up MixMonitor on the calling channel before doing the Dial. The call sounds fine to both parties in the call, but when listening back to the recording, the speech is slowed down and broken - a bit like a robot or dalek voice. Examining the recorded audio in Goldwave I can see that every 20ms (the size of a SIP packet), an extra 20ms of silence is being inserted in the recording file. This happens most of the time, punctuated by occasional bursts of clear audio before it starts happening again. Has anyone seen this kind of thing before? Better still, seen it and solved it? The SIP phones are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony Maybe a I'm reaching here but.. I had some very strange issues with broken quality with Monitor and a NFS mount. This was 1.4, but several years ago. I ended up not using NFS in the end. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI Permissions, all means different things?
Hi! I'm trying to limit the permissions for a AMI-account. But I'm a little bit confused by the permissions. The commands I use are (output from manager show commands, btw: privilege col seems cropped?): Action PrivilegeSynopsis Redirect call,all Redirect (transfer) a call. Originateoriginate,allOriginate a call. Getvar call,reporting, Gets a channel variable. If I put this in my manager.conf: [pbx_ami] secret = *** deny=0.0.0.0/0.0.0.0 permit = x.x.x.x/255.255.255.255 write=originate,call read= I get this (manager show user pbx_ami): username: pbx_ami secret: Set acl: yes read perm: none write perm: call,originate,all displayconnects: yes Where does the all permission come from? However, If I change the row in manager.conf to write=originate,call,all the output is: username: pbx_ami secret: Set acl: yes read perm: none write perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,all displayconnects: yes Can someone explain this please? Thanks! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do you convert your prompts to an asterisk-friendly format?
Hi,
I've used the shells-script at the end of this email to generate 8khz
mono wave-files for asterisk from a 144 khz recording.
The script does two things: resample normalize the audio volume.
Anyone like to share their recommendations / scripts for doing this
conversion? I've just converted to 8khz wave, should I convert to
something else?
For the googler in the future this is my current script (which I hope to
improve):
BASEDIR=`dirname $0`
PROMPTDIRS=dir1 dir2
for dir in ${PROMPTDIRS}
do
src=${BASEDIR}/recordings/prompts/${dir}
dst=${BASEDIR}/generated/prompts/8khz/${dir}
for i in ${src}/*.wav; do sox $i -V -r 8000 -c 1 -q -s \
${dst}/$(basename $i .wav).wav vol 0.8; done
normalize-audio -a -20dBFS ${dst}/*
done
--
Johan Wilfer
JT Technologies Telecommunications AB
Jabber: jo...@jttech.se | Phone: +46 31 3809100
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Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?
2012-08-28 16:44, Andrew Latham skrev: On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote: Hi, I've used the shells-script at the end of this email to generate 8khz mono wave-files for asterisk from a 144 khz recording. Try this to test with http://www.digium.com/en/products/ivr/audio-converter.php and compare your output first... Interesting. Didn't know about this. It's good for testing, but I would like to automate it. Is the source-code open or available? -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Phone: +46 31 3809100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?
2012-08-28 17:04, Andrew Latham skrev: Yep, check out repotools for that http://svn.asterisk.org/svn/repotools/sound_tools/scripts/ Cool! Thank you! -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Phone: +46 31 3809100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One leg in a conference and adjusting stream volume of other leg
2012-08-27 19:48, Markus skrev: Hi Matthew, Am 27.08.2012 15:41, schrieb Matthew Jordan: When they adjust the volume of the stream, if effects only their stream, and not the volume of the stream of the other callers. In short: All callers at all times are *always* in the same conference, but each caller is able to increase or decrease the volume of their MP3 stream individually. You can use ConfBridge's DTMF menus to allow a participant to change their listening volume. This should only affect the audio that the participant hears, and not the other participants in the conference - regardless of whether or not the audio originates from the same source. thanks! I wasn't clear enough in my original mail. What I meant is: the volume of the stream that a user is listening to is adjusted, but the volume of the conference itself is not changed! That means, a conference is going on, and everyone is listening to the same music at the same time, but when the music becomes too loud or annoying, a user can individually adjust the volume of his music, while the volume of the speech of each user, basically the conference itself, remains the same. I think what I'm looking for is to inject the MP3 stream into only the listening direction of each user, and allow its volume to get adjusted via DTMF. And at the same time, each user is in the same conference. Even more: I would like to be able to feed each user a *different* volume-adjustable MP3 stream, but all of the users are still in the same conference (not hearing each others MP3 stream, only their voice!). I've researched high and low and came up with the following pointers: - Dial with the G flag - ChanSpy, whispering - VOLUME() - MOH connected to a local channel - Queue that loops indefinitely But I don't know yet how to put it all together. I found some hints in the right direction here: Playing audio to one channel only: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg245811.html Meetme with background music (last post) http://fonality.com/trixbox/forums/trixbox-forums/help/meetme-background-music Background music during a call http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html Does anyone have the right solution and is available to create a dialplan for me for cash? Please get in touch! I would do it like this: 1. Use Meetme or Confbridge and use functionality to jump out of the conference if DTMF is pressed (X-flag in meetme, I expect similar exists in confbridge). 2. Call AGI, Log to DB, etc - whatever - and return to the conference. 3. Have a external program that manipulates the channel playing the music. For example this could be done by ChannelRedirect AMI to special dialplan extensions that lower and raises the volum. You can use System()-app in asterisk, or AGI for example. Then use AMI in the script. The music on hold could be implemented as a Local channel. a. Look at Originate-app, or Originate AMI command. One side of the call are connected to a context/extension/priority (for example: Meetme here). And the other end you dial Local/extension@context (for example: Here you play music). b. Prepare some extensions that lower/raises volume (look at func_volume) Good luck! /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UDP miss a hangup on SIP
2012-08-16 02:13, Jerry Geis skrev: Is it possible to miss a UDP SIP packet to hangup a call? Using 1.4.43 I had a call from on asterisk box (server) to a low end client (chan_alsa) not hangup. Could this be due to missed UDP SIP packet to hangup? Is there anyway for a client asterisk (chan_alsa again) to monitor the connection and if the channel is not there to hangup? In sip.conf you could use rtp-timers to hangup a call if the media-stream stops to flow. Look at these options in sip.conf: rtptimeout=60 rtpholdtimeout=300 rtpkeepalive=0 -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 22:32, Johan Wilfer skrev: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! I contacted them and will use a server connected to a switch-port in mirroring mode. The gui seems like a great tool in troubleshooting. Nobody uses the rtcp-stats in asterisk for quality monitoring? Other suggestions? -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating different options. Aside from the manual thing I could see two variants: - Dump the traffic (on the server or another via switch port mirroring/monitoring) and analyze it with tshark - Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking for some ideas to setup this so I can eliminate this manual and time-consuming process in the future. And know about the problems before the customer complains about the quality.. Thanks in advance! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Define custom vm-login sound file per VM context?
2012-02-21 19:20, Todd Routhier skrev: OK, this will work and is probably a better solution than the language idea. Although, the language idea just sounds easier and a little more fun :-) Hmm, I think I will try the language solution and see if it works with a fake country/language code like Cust327 or whatever. Just wonder if that will break anything else now or with future upgrades. Thanks for all the help! You can also use en_baselevel_customer234 as a language, asterisk will first try to find a soundfile in the en_baselevel_customer234-directory, and if not found in the en_baselevel-directory. After that it will look in the en-dir. Can't find the docs for this right now but this way you don't need to copy all the recordings, and you can stack as many layers as you like.. :-) /Johan -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound file
2012-01-26 10:11, Eyal skrev:
Thanks
But this is not what I am looking for, in this way I can start the
sound file from some point in the file but the callers must hear the
file until the end.
I need something that allows me to start from some place in the file
and end it in some other place in the file (say song from time 01:32
until 01:57),
Or
Like the *controlplayback* doing fast-forward but without having to
click any key by caller.
You can do that by combining ControlPlayback and use TIMEOUT function.
If you don't want the user to be able to use any keys you can use all
keys as stop-keys in ControlPlayback and have some logic restart the
playback at position ${CPLAYBACKOFFSET}.
For more details do:
core show application ControlPlayback
core show function TIMEOUT
/Johan
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nasir
Iqbal
*Sent:* Thursday, January 26, 2012 10:53 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] play sound file
check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
Nasir Iqbal
ICTBroadcast
SMS, Fax and Voice broadcasting solution
http://www.ictbroadcast.com/
On Wed, Jan 25, 2012 at 8:29 PM, Eyal e...@mcr-m.com
mailto:e...@mcr-m.com wrote:
Hi,
How can I play a sound file from the middle and end it after a certain
number of seconds?
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JT Tech | Developer webb: http://jttech.se
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Re: [asterisk-users] meetme - Unable to write frame to channel
2012-01-20 20:09, Matt Hamilton skrev: Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? As you write I have seen this also with SIP in Meetme conferences sometimes when sip-channels is hung up. I havn't found any real problem or bad sound related to this, so I usually ignore this error. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-20 00:38, John Knight skrev: Why doesn't Debian use the rhel6-openvz-kernel if that is the one that is maintained? Are you sure they use an outdated kernel? I didn't see an el6 tag on your kernel version that you first posted which means it's probably based on the 2.6.32 vanilla branch (http://wiki.openvz.org/Download/kernel/2.6.32). The el6 version (http://wiki.openvz.org/Download/kernel/rhel6) is the only known actively developed branch of 2.6.32 that I know of. I can't imagine Debian not packaging it up correctly by not reflecting the correct branch information. Though it's only been a year and half since 2.6.32-el6 has been around, I've already seen quite a bit of bug fixes, security fixes and backports added to it so it's already diverging quite heavily from the vanilla branch. Something that works flawlessly on 2.6.32-el6 might not work the same way on 2.6.32, and I'm wondering if this might be a cause of issues. I did some reading on the openvz forum / wiki. They seems to also update and fix the debian branch from what I found. Anyway I don't have the time to build this server from scratch anyway so it will have to stay the way it is unless it's seriously broken. For the next server I will research will further, and test out lxc to see if it can replace openvz. I found some posts to this mailinglist about lxc that seems to indicate that it works well, and the methods are quite similar to openvz. To Digium: Does Digium test dahdi against a specific set of kernels such as 2.6.18-el5 and 2.6.32-el6 or do they only test against the vanilla upstream branches or a mixture? Dahdi target platforms would be interesting to know in relation to the context of Johan's dahdi problem. Maybe I will try switch to lxc instead of openvz as it is in the mainline kernel now. After all I need two things: Isolation, and the possibility to run multiple asterisk VEs on the same physical machine. Hmm, I've never used lxc. It definitely sounds interesting. If you're going to implement it running Asterisk, I'd love to know if there are any issues or special methods required to get dahdi running (such as the DEVNODES feature in vz). Also, sorry to anyone if I've veered too far offtopic, I'm quite interested and invested in openvz/asterisk/dahdi interoperability. I'll keep you updated! Thanks for you input! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-19 13:27, Doug Lytle skrev: Tony Mountifield wrote: It may be a stupid question just displaying ignorance on my part, but why are you using*AMD*64 architecture on an *Intel* processor? Surely for 64-bit, you should be using x86_64 architecture instead? From what I've read, AMD came out with the extended instruction set and Intel just implemented it as is. It's basically the same for both processors. Doug I found this wiki http://wiki.debian.org/DebianAMD64Faq , it states that AMD64 / Intel64 / x86_64 are the same thing: Debian on AMD64 FAQ Q: Is this port only for AMD 64-bit CPUs? A: No. AMD64 is the name chosen by AMD for their 64-bit extension to the Intel x86 instruction set. Before release, it was called x86-64 or x86_64, and some distributions still use these names. Intel refers to its AMD64 implementation as Intel64 previously named EM64T. The architecture is AMD64-compatible and Debian AMD64 will run on AMD and Intel processors with 64-bit support. Because of the technology paternity, Debian uses the name AMD64. I've always wondered about the amd in the name, but it makes sense now. Thanks for the input.. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 20:06, Shaun Ruffell skrev: That's pretty severe, and could certainly cause problems for DAHDI trying to use the kernel as a timing source. NTP will correct the drift, but the drift is still happening and it's not corrected on every tick. If the ticks are not happening at the rate they are supposed to, then DAHDI will not be operating at the clock rate it is supposed to. Kevin, this looks like a good candidate for using the monotonic interface in the kernel that we were talking about last week or the week before. The specific function call escapes me at the moment. Johan, I can't do it right this second, but I'll prepare an issue / patch against a 2.6.32 kernel that should make dahdi less prone to clock skew from NTP (although you probably want to get that fixed somehow) if you would be willing to test it for me on your server. Another thing you can try in the meantime is switch to DAHDI 2.5.0.2 and edit drivers/dahdi/Kbuild to enable dahdi_dummy which will use the (relatively inefficient for the purposes of conferencing) highres timers when loaded by default on recent kernels (if compiled in). Yes! That did it. Thank you! This time it wasn't as much channels active as the last time. It was 32 channels active, and the last time 57. In the beginning of the next week there will be more users to reproduce the test with more channels, and I will begin to work with a test configuration to create some channels from another server. (Should have done that a long time ago I guess) I will hold up migration of more customers to this server, so we can test your patch against Dahdi 2.6. This is the the server with dahdi_dummy idle: 99.999% 99.997% 100.000% 99.999% 99.999% 99.994% 99.997% 99.999% 99.997% 99.999% 99.999% 99.999% 99.999% 99.999% 99.998% 99.999% 99.999% 99.998% 99.998% 99.999% 99.999% 99.999% 99.998% 99.999% 99.999% 99.998% 99.998% 100.000% 99.999% 99.998% 99.999% 99.999% 99.998% 99.998% 99.999% 99.998% 99.998% 99.999% 99.998% ^C --- Results after 10335 passes --- Best: 100.000 -- Worst: 99.984 -- Average: 99.998134, Difference: 99.998607 This is the server with dahdi_dummy during load: 99.996% 99.999% 99.998% 99.992% 99.997% 99.999% 99.996% 99.993% 99.999% 99.998% 99.992% 99.994% 99.996% 99.992% 99.994% 99.996% 99.998% 99.998% 99.998% 99.997% 99.999% 99.995% 99.993% 99.998% 99.997% 99.998% 99.996% 99.995% 99.998% 99.990% 99.996% 99.996% 99.997% 99.992% 99.994% 99.986% 99.998% 99.991% 99.987% 99.991% 99.999% ^C --- Results after 201 passes --- Best: 100.000 -- Worst: 99.962 -- Average: 99.994760, Difference: 99.998428 Thanks! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 19:44, John Knight skrev: Have you used 64 bit kernels (amd64) in your setup? Distribution? Aye, I use the current stable 64-bit rhel6 branch openvz kernel with centos 6 on the node and scientific linux 6 in the template without issue other than what I described before with res_timing_timerfd.so pegging the cpu and coring Asterisk. Good to know that it is working! I've run i386 earlier, but thought it was time to try 64-bit. It's never a suggestion a debian user wants to hear, but as the vanilla 2.6.32 openvz kernel has effectively been abandoned by the OpenVZ dev team in favor of the rhel6 version of 2.6.32, and since the node shouldn't really be doing anything other than hosting the templates, have you considered running centos6/rhel6-openvz kernel on the node and debian in the containers? Just a suggestion, but no further openvz development is being done to the vanilla 2.6.32 branch and the rhel6 openvz kernel will consistently have bug fixes and and backports. Not trying to start a distro war or anything, rather just a suggestion. I've been quite happy with Debian. Previously I was using BSD, and it was almost impossible to upgrade the system. And apt / dpkg have never failed me, very impressive. I guess rhel works well also, but I've little experience with it. Why doesn't Debian use the rhel6-openvz-kernel if that is the one that is maintained? Are you sure they use an outdated kernel? I have to read up on this, the next server maybe should use another distro for the HN. Maybe I will try switch to lxc instead of openvz as it is in the mainline kernel now. After all I need two things: Isolation, and the possibility to run multiple asterisk VEs on the same physical machine. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi for meetme on AMD64 arch?
I'm in the process of replacing an old server with a new one and are making som changes in the infrastructure, the biggest change in my eyes is moving from i386 to AMD64 arch. Yesterday I began migrating some users from the old to the new server. After only 57 concurrent calls in abount 13 conferences the sound are losing quality. The server uses dahdi 2.6.0 for timing but no dahdi hardware. dahdi_test gives results like this when the server is used like that: 100.000% 99.999% 99.994% 99.998% 99.999% 99.616% 99.614% 99.997% 99.998% 99.618% 99.615% 99.994% 99.987% 99.626% 99.628% 99.993% 99.626% 100.000% 100.000% 99.622% 99.999% 99.607% 99.604% 99.627% 99.621% 99.629% 99.627% 99.998% 99.622% 99.995% 99.621% 99.996% Results from dahdi_test with only some calls active: 99.999% 99.999% 99.990% 99.998% 99.999% 99.995% 99.995% 99.993% 99.997% 99.993% 99.999% 99.998% 99.996% 99.996% 99.998% 99.998% 99.991% 99.998% 99.995% 99.995% 99.987% 99.985% 99.996% 99.995% Looking at the cacti graphs the kernel uses 100% cpu (total 400% with 4 processor cores), when the problem above is present. Top does not show this kernel-cpu that cacti shows, but this maybe is by design? Asterisk is using about 15% cpu. top - 19:32:06 up 20:57, 1 user, load average: 0.00, 0.00, 0.00 Tasks: 213 total, 1 running, 212 sleeping, 0 stopped, 0 zombie Cpu(s): 7.4%us, 29.6%sy, 0.0%ni, 55.3%id, 0.0%wa, 0.0%hi, 7.7%si, 0.0%st Mem: 12299332k total, 3967800k used, 8331532k free, 251432k buffers Swap: 19529720k total,0k used, 19529720k free, 2919456k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30666 root 0 -20 539m 25m 6600 S 15 0.2 6:55.01 asterisk 738 root 20 0 19184 1444 1004 R1 0.0 0:00.08 top The old server (i386 Debian 5: Linux 2.6.26-2-openvz-686) can have 320 calls in conferences without this problem. The new server (amd64 Debian 6: Linux 2.6.32-5-openvz-amd64) show these problems after 50 calls.. Old server: Hp dl360g5, 4 cpu Xeon E5420, 2.50GHz run i386 with PAE and OpenVZ, Debian Lenny uses the broadcom nic's on the motherboard asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti shows cpu in kernel mode 80% with 320 active calls in conferences New server: Hp dl360g7, 4 cpu Xeon E5520, 2.27GHz run amd63 with OpenVZ, Debian Squeeze uses Intel nic's 82571EB for offloading the processor + nic bonding in the kernel for failover. asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti show cpu in kernel mod 100% with 57 active calls in conferences This is a puzzle to me.. - Does anyone have experience with amd64 arch and dahdi for timing? - Can Dahdi om amd64 be responsible for the high cpu in kernel mode? - I have a spare Digium TE220, would it offload the server to use it as a timing source only? - How do I debug the high cpu usage by the kernel, can I break this down by module in some way? Many, many thanks! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 11:31, John Knight skrev: Hi Johan, I've run into a similar issue before. I didn't resolve the problem per se, but I got around it by modifying modules.conf to disable the loading of res_timing_timerfd.so and loaded res_timing_dahdi.so instead: noload = res_timing_timerfd.so load = res_timing_dahdi.so Cpu load came back down and call quality has been excellent since. Perhaps this might work for you? Hi! I think the timing support was included in asterisk in 1.6.1/1.6.2. As I run 1.4 these modules are not available at all. Do you run asterisk 1.6 and amd64? Another option would be to port my dialplan to a newer version of asterisk if this can resolve the issue. A workaround I've been tinking about is to try to put a spare Digium-card in the server just for timing, if there is something strange with the soft dahdi timing. I'm not very fond of the idea to rebuild everything on i386 architecture, but that's the last resort. /Johan On 1/18/2012 4:24 AM, Johan Wilfer wrote: I'm in the process of replacing an old server with a new one and are making som changes in the infrastructure, the biggest change in my eyes is moving from i386 to AMD64 arch. Yesterday I began migrating some users from the old to the new server. After only 57 concurrent calls in abount 13 conferences the sound are losing quality. The server uses dahdi 2.6.0 for timing but no dahdi hardware. dahdi_test gives results like this when the server is used like that: 100.000% 99.999% 99.994% 99.998% 99.999% 99.616% 99.614% 99.997% 99.998% 99.618% 99.615% 99.994% 99.987% 99.626% 99.628% 99.993% 99.626% 100.000% 100.000% 99.622% 99.999% 99.607% 99.604% 99.627% 99.621% 99.629% 99.627% 99.998% 99.622% 99.995% 99.621% 99.996% Results from dahdi_test with only some calls active: 99.999% 99.999% 99.990% 99.998% 99.999% 99.995% 99.995% 99.993% 99.997% 99.993% 99.999% 99.998% 99.996% 99.996% 99.998% 99.998% 99.991% 99.998% 99.995% 99.995% 99.987% 99.985% 99.996% 99.995% Looking at the cacti graphs the kernel uses 100% cpu (total 400% with 4 processor cores), when the problem above is present. Top does not show this kernel-cpu that cacti shows, but this maybe is by design? Asterisk is using about 15% cpu. top - 19:32:06 up 20:57, 1 user, load average: 0.00, 0.00, 0.00 Tasks: 213 total, 1 running, 212 sleeping, 0 stopped, 0 zombie Cpu(s): 7.4%us, 29.6%sy, 0.0%ni, 55.3%id, 0.0%wa, 0.0%hi, 7.7%si, 0.0%st Mem: 12299332k total, 3967800k used, 8331532k free, 251432k buffers Swap: 19529720k total,0k used, 19529720k free, 2919456k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30666 root 0 -20 539m 25m 6600 S 15 0.2 6:55.01 asterisk 738 root 20 0 19184 1444 1004 R1 0.0 0:00.08 top The old server (i386 Debian 5: Linux 2.6.26-2-openvz-686) can have 320 calls in conferences without this problem. The new server (amd64 Debian 6: Linux 2.6.32-5-openvz-amd64) show these problems after 50 calls.. Old server: Hp dl360g5, 4 cpu Xeon E5420, 2.50GHz run i386 with PAE and OpenVZ, Debian Lenny uses the broadcom nic's on the motherboard asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti shows cpu in kernel mode 80% with 320 active calls in conferences New server: Hp dl360g7, 4 cpu Xeon E5520, 2.27GHz run amd63 with OpenVZ, Debian Squeeze uses Intel nic's 82571EB for offloading the processor + nic bonding in the kernel for failover. asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti show cpu in kernel mod 100% with 57 active calls in conferences This is a puzzle to me.. - Does anyone have experience with amd64 arch and dahdi for timing? - Can Dahdi om amd64 be responsible for the high cpu in kernel mode? - I have a spare Digium TE220, would it offload the server to use it as a timing source only? - How do I debug the high cpu usage by the kernel, can I break this down by module in some way? Many, many thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 16:45, John Knight skrev: Ah, apologies, I just re-read your given Asterisk version. Indeed, I was using 1.8.5.0 at the time, not any 1.4.x release. Any digium timing card will work as an OpenVZ compatible dahdi timing device, I've seen this work on both Virtuozzo and OpenVZ. Setting it up, there's no difference in how you set up passthrough access using DEVNODES to the device from /dev inside the $CTID.conf file. Just make sure permissions inside the container make it writable by the asterisk user. Okay, I will try that procedure tonight. I'll also remove my Intel dual nic card, and the network bonds. After that, then only difference to the machine working and the machine not working are i386 / amd6. And the os version - debian 5 / 6. Have you used 64 bit kernels (amd64) in your setup? Distribution? Thanks for your advices, it's very appreciated! /Johan On 1/18/2012 8:52 AM, Johan Wilfer wrote: 2012-01-18 11:31, John Knight skrev: Hi Johan, I've run into a similar issue before. I didn't resolve the problem per se, but I got around it by modifying modules.conf to disable the loading of res_timing_timerfd.so and loaded res_timing_dahdi.so instead: noload = res_timing_timerfd.so load = res_timing_dahdi.so Cpu load came back down and call quality has been excellent since. Perhaps this might work for you? Hi! I think the timing support was included in asterisk in 1.6.1/1.6.2. As I run 1.4 these modules are not available at all. Do you run asterisk 1.6 and amd64? Another option would be to port my dialplan to a newer version of asterisk if this can resolve the issue. A workaround I've been tinking about is to try to put a spare Digium-card in the server just for timing, if there is something strange with the soft dahdi timing. I'm not very fond of the idea to rebuild everything on i386 architecture, but that's the last resort. /Johan On 1/18/2012 4:24 AM, Johan Wilfer wrote: I'm in the process of replacing an old server with a new one and are making som changes in the infrastructure, the biggest change in my eyes is moving from i386 to AMD64 arch. Yesterday I began migrating some users from the old to the new server. After only 57 concurrent calls in abount 13 conferences the sound are losing quality. The server uses dahdi 2.6.0 for timing but no dahdi hardware. dahdi_test gives results like this when the server is used like that: 100.000% 99.999% 99.994% 99.998% 99.999% 99.616% 99.614% 99.997% 99.998% 99.618% 99.615% 99.994% 99.987% 99.626% 99.628% 99.993% 99.626% 100.000% 100.000% 99.622% 99.999% 99.607% 99.604% 99.627% 99.621% 99.629% 99.627% 99.998% 99.622% 99.995% 99.621% 99.996% Results from dahdi_test with only some calls active: 99.999% 99.999% 99.990% 99.998% 99.999% 99.995% 99.995% 99.993% 99.997% 99.993% 99.999% 99.998% 99.996% 99.996% 99.998% 99.998% 99.991% 99.998% 99.995% 99.995% 99.987% 99.985% 99.996% 99.995% Looking at the cacti graphs the kernel uses 100% cpu (total 400% with 4 processor cores), when the problem above is present. Top does not show this kernel-cpu that cacti shows, but this maybe is by design? Asterisk is using about 15% cpu. top - 19:32:06 up 20:57, 1 user, load average: 0.00, 0.00, 0.00 Tasks: 213 total, 1 running, 212 sleeping, 0 stopped, 0 zombie Cpu(s): 7.4%us, 29.6%sy, 0.0%ni, 55.3%id, 0.0%wa, 0.0%hi, 7.7%si, 0.0%st Mem: 12299332k total, 3967800k used, 8331532k free, 251432k buffers Swap: 19529720k total,0k used, 19529720k free, 2919456k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30666 root 0 -20 539m 25m 6600 S 15 0.2 6:55.01 asterisk 738 root 20 0 19184 1444 1004 R1 0.0 0:00.08 top The old server (i386 Debian 5: Linux 2.6.26-2-openvz-686) can have 320 calls in conferences without this problem. The new server (amd64 Debian 6: Linux 2.6.32-5-openvz-amd64) show these problems after 50 calls.. Old server: Hp dl360g5, 4 cpu Xeon E5420, 2.50GHz run i386 with PAE and OpenVZ, Debian Lenny uses the broadcom nic's on the motherboard asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti shows cpu in kernel mode 80% with 320 active calls in conferences New server: Hp dl360g7, 4 cpu Xeon E5520, 2.27GHz run amd63 with OpenVZ, Debian Squeeze uses Intel nic's 82571EB for offloading the processor + nic bonding in the kernel for failover. asterisk 1.4.42 in openvz container (uses /dev/dahdi for timing) cacti show cpu in kernel mod 100% with 57 active calls in conferences This is a puzzle to me.. - Does anyone have experience with amd64 arch and dahdi for timing? - Can Dahdi om amd64 be responsible for the high cpu in kernel mode? - I have a spare Digium TE220, would it offload the server to use it as a timing source only? - How do I debug the high cpu usage by the kernel, can I break this down by module in some way? Many, many thanks
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 17:50, Shaun Ruffell skrev: On Wed, Jan 18, 2012 at 02:52:47PM +0100, Johan Wilfer wrote: 2012-01-18 11:31, John Knight skrev: Hi Johan, I've run into a similar issue before. I didn't resolve the problem per se, but I got around it by modifying modules.conf to disable the loading of res_timing_timerfd.so and loaded res_timing_dahdi.so instead: noload = res_timing_timerfd.so load = res_timing_dahdi.so Cpu load came back down and call quality has been excellent since. Perhaps this might work for you? Hi! I think the timing support was included in asterisk in 1.6.1/1.6.2. As I run 1.4 these modules are not available at all. Do you run asterisk 1.6 and amd64? Another option would be to port my dialplan to a newer version of asterisk if this can resolve the issue. A workaround I've been tinking about is to try to put a spare Digium-card in the server just for timing, if there is something strange with the soft dahdi timing. I would be interested to learn if there was any problem with soft (coretimer) when DAHDI is running on your ne platform. I would not expect that. One question first though, is your new server able to keep accurate time with nt, or is the clock drifting or experiencing heavy jitter? The clock is accurate by ntp sync. It uses the vanilla debian config you get if you apt-get install ntp. Was it nt(p) you meant above? The clock drifts a lot if it is not synced by ntp. I've noticed most of my hp 360/380 servers to drift up to 10 minutes per week, including this server. But ntp fixes this right? If you have ideas how to debug this I would be very grateful. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi for meetme on AMD64 arch?
2012-01-18 20:06, Shaun Ruffell skrev: On Wed, Jan 18, 2012 at 12:58:31PM -0600, Kevin P. Fleming wrote: On 01/18/2012 12:15 PM, Johan Wilfer wrote: 2012-01-18 17:50, Shaun Ruffell skrev: One question first though, is your new server able to keep accurate time with nt, or is the clock drifting or experiencing heavy jitter? The clock is accurate by ntp sync. It uses the vanilla debian config you get if you apt-get install ntp. Was it nt(p) you meant above? The clock drifts a lot if it is not synced by ntp. I've noticed most of my hp 360/380 servers to drift up to 10 minutes per week, including this server. But ntp fixes this right? That's pretty severe, and could certainly cause problems for DAHDI trying to use the kernel as a timing source. NTP will correct the drift, but the drift is still happening and it's not corrected on every tick. If the ticks are not happening at the rate they are supposed to, then DAHDI will not be operating at the clock rate it is supposed to. Didn't think of that. I've turnd of ntpd now to see exactly how much the clock skew when ntpd is not running. root@milkyway:/home/johan# ntptime ntp_gettime() returns code 0 (OK) time d2c1ac73.45214000 Wed, Jan 18 2012 21:39:15.270, (.270039), maximum error 167603 us, estimated error 815 us ntp_adjtime() returns code 0 (OK) modes 0x0 (), offset 522.000 us, frequency 124.445 ppm, interval 1 s, maximum error 167603 us, estimated error 815 us, status 0x1 (PLL), time constant 6, precision 1.000 us, tolerance 500 ppm, This is the server running dahdi_test at the same time: root@milkyway:/home/johan# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.602% 99.991% 99.614% 99.983% 99.608% 99.999% 99.611% 100.000% 99.999% 99.998% 99.999% 99.995% 99.609% 99.613% 99.997% 99.608% 99.611% 99.999% 99.608% 99.610% 99.998% 99.999% 99.999% 99.995% 99.987% 99.999% 99.999% 99.999% 99.999% 99.996% 99.999% 99.999% 99.995% 99.998% 99.999% 99.999% 99.999% 99.998% 99.999% 99.992% 99.994% 99.999% 99.989% 99.999% 99.998% 99.998% 99.996% 99.998% 99.998% 99.983% 99.999% 99.998% 99.999% 99.992% 99.997% 99.997% 99.982% 99.979% 99.986% 99.993% 99.999% 99.999% 99.999% 99.995% 99.999% 99.997% 99.993% 99.995% 99.998% 99.998% 99.999% 99.998% 99.998% 99.998% 99.999% 99.999% 99.999% ^C --- Results after 77 passes --- Best: 100.000% -- Worst: 99.602% -- Average: 99.945879% Cummulative Accuracy (not per pass): 99.998 Kevin, this looks like a good candidate for using the monotonic interface in the kernel that we were talking about last week or the week before. The specific function call escapes me at the moment. Johan, I can't do it right this second, but I'll prepare an issue / patch against a 2.6.32 kernel that should make dahdi less prone to clock skew from NTP (although you probably want to get that fixed somehow) if you would be willing to test it for me on your server. Another thing you can try in the meantime is switch to DAHDI 2.5.0.2 and edit drivers/dahdi/Kbuild to enable dahdi_dummy which will use the (relatively inefficient for the purposes of conferencing) highres timers when loaded by default on recent kernels (if compiled in). Cheers, Shaun Great, I will try this. I start with dahdi_dummy and check the results. Thank you! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
2012-01-16 19:41, asterisk jobs skrev: Hello, I can do simple, yum install asterisk18-* and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, /*You do not appear to have the source for the 2.6.32-4-pve kernel installed.*/ /* */ 1- Based on above error and Google search I have concluded that dahdi-linux module should be installed on mother node. So, I am puzzled. How does Digium yum repository achive this without acessing the mother node? 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? I've just installed a new server with OpenVZ. And as others has explained you will need Dahdi for Meetme among other things. You will need to install dahdi-complete on the Hardware node, and the kernel sources. ( Debian: apt-get install linux-headers-`uname -r` ) In the VE conf-file you will need the following line for the VE to access Dahdi: DEVNODES=dahdi/channel:rw dahdi/ctl:rw dahdi/pseudo:rw dahdi/timer:rw In the VE, compile and install dahdi-complete, then build and install asterisk. -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] State of Asterisk+Virtualization+Timing
2011-11-01 18:08, Tim Nelson skrev: Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent issues that need to be addressed: OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant access to host node timing source (physical device, or dahdi_dummy in /dev/dahdi/) to the containerized Asterisk process. KVM - Higher overhead, easier installation, 'true virtualization'. Primary issue is not timing per se, but KVM scheduling. Timing source, while present from dahdi_dummy natively may still not get proper scheduling by KVM process. This could also affect general call quality (even non IAX2 trunked voice), DTMF, etc. I have to believe there are others running virtualized Asterisk installations with some degree of success on OpenVZ or KVM. Care to share your thoughts? Hi Tim, I'm using OpenVZ, it works very well. Take a look at: http://wiki.openvz.org/Asterisk_from_source You will have to compile dahdi and install it on the HN, and then you compile and install asterisk in the containers. I've not done this with chan_iax, but with meetme. In the case with meetme you would have to move some files over to trick asterisk that dahdi is compiled on the machine. The wiki mentions copying user.h. I used this as a starting point, some years ago: http://www.telephreak.org/papers/vpa/ This paper covers vserver, so it's not exactly the same - but the steps with tonezone was the same when I built the current server running this configuration. I'm in the process of building another server with openvz, so I'll need to refresh my memory and try to document the procedure. -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk Sessions on same machine
On 2011-07-19 20:07, Michael wrote: Hello, We would like Asterisk to listen on port 5060 and on an additional port. From what we read online, it's not really possible, so is it possible to install a separate instance of Asterisk on the same machine (without using Vmware or such) and set the 2nd instance to listen on another port? I've used openvz for this, it's not realy virtualisation - all the virtual machines (asterisk-boxes) share the same linux kernel. If you want to use dahdi/meetme you will have to let the VE's use the /dev/dahdi devices but thats not very hard. -- Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stream rtp from asterisk
On 2011-07-04 15:07, Marcus Kvarsell wrote: Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms Asterisk patch. See here. If there is any folk out there that has knowledge of this or any similar software I would be very happy if you could help me get this to work. / Marcus http://oreka.sourceforge.net/oreka-user-manual.html#gettingvoiptraffic Seems they propose setting the switch in mirror/monitoring-mode and sniff the traffic on another server. Normal managed and smart-switches support this option... Or you can install the software on the asterisk server. /Johan -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov Skickat: den 4 juli 2011 14:49 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] stream rtp from asterisk On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Re: ControlPlayback's options
On 2011-06-10 07:30, virendra bhati wrote: Hi John, Sorry for wrong information. Actually it's J not P option in ControlPlayBack... http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback That page is correct for asterisk 1.4 but the feature you need is in 1.6.0 and forward. Have you checked to documentation in the CLI? The J option is something different.. /Johan On Fri, Jun 10, 2011 at 12:23 AM, Johan Wilfer li...@jttech.se mailto:li...@jttech.se wrote: Humm... Seems like my message didn't make it. Here we go again.. /Johan Original Message Subject:Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer li...@jttech.se mailto:li...@jttech.se To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com On 2011-06-05 19:54, virendra bhati wrote: Hi John Wilfer, Thanks for replay. Now all things is working on asterisk 1.6.2.18 version. But When I try the same thing on Asterisk 1.4.X then facing problem. Is this the problem of ControlPlayback 's option fields of asterisk 1.4.X in this version have option P(jumping) not O(time) ? Is there any way by which we will implement like by upload ControlPlayback from asterisk 1.6 to 1.4 or else ? ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]]) These features only exist in 1.6.0 and forward. Should be relative easy to make a backport if you need to run it on 1.4 I've never heard of the P()-option, which version of asterisk? /Johan -- Med vänlig hälsning Johan Wilfer email:jo...@jttech.se mailto:jo...@jttech.se JT Tech | Utvecklare webb:http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Re: ControlPlayback's options
Humm... Seems like my message didn't make it. Here we go again.. /Johan Original Message Subject:Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer li...@jttech.se To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On 2011-06-05 19:54, virendra bhati wrote: Hi John Wilfer, Thanks for replay. Now all things is working on asterisk 1.6.2.18 version. But When I try the same thing on Asterisk 1.4.X then facing problem. Is this the problem of ControlPlayback 's option fields of asterisk 1.4.X in this version have option P(jumping) not O(time) ? Is there any way by which we will implement like by upload ControlPlayback from asterisk 1.6 to 1.4 or else ? ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]]) These features only exist in 1.6.0 and forward. Should be relative easy to make a backport if you need to run it on 1.4 I've never heard of the P()-option, which version of asterisk? /Johan -- Med vänlig hälsning Johan Wilfer email:jo...@jttech.se JT Tech | Utvecklare webb:http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ControlPlayback's options
On 2011-06-04 13:38, virendra bhati wrote:
Hi Johan Wilfer,
Thanks for your reply. On the basis of your provided code I made all
things into extensions.conf. But i have an small issue on which I need
your attention again.
in below context what's ${tz} ? Is this time zone value or else?
Yes, I store the calls timezone in tis variable before the code sample
you got.
and another things is what is the use of SayUnixTime(${time},${tz},d
'digits/of' B);
this function in such case?
I've implemented 5 as a pause-button on the phone. This context handles
this by playing a
prompt that you have pause the recording and time and date.
This is repeated untill the user presses a key on the keypad.
context conference_play_recordings_
conference_paused {
announce = {
Set(time=$[${epoch_start}+${position}/1000]);
while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
}
}
one thing which is also confusing is that what is the meaning or use
of such lines in this application.
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))
${filename} is the file you want to play.
6 is 60 seconds to skip.
3 is to use 3 as forward 60 seconds
1 to to use 1 as rewind 60 seconds
*#2456790 is used as stop buttons (and handled by the dialplan)
o() is a option to go to a specific position in the file
${position} is the variable that hold the current position of the playback.
To get more details use the following command:
asterisk*CLI core show application ControlPlayback
Displays:
-= Info about application 'ControlPlayback' =-
[Synopsis]
Play a file with fast forward and rewind.
[Description]
This application will play back the given filename.
It sets the following channel variables upon completion:
${CPLAYBACKSTATUS}: Contains the status of the attempt as a text string
SUCCESS
USERSTOPPED
ERROR
${CPLAYBACKOFFSET}: Contains the offset in ms into the file where playback
was at when it stopped. '-1' is end of file.
${CPLAYBACKSTOPKEY}: If the playback is stopped by the user this variable
contains the key that was pressed.
[Syntax]
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
[Arguments]
skipms
This is number of milliseconds to skip when rewinding or fast-fo
rwarding.
ff
Fast-forward when this DTMF digit is received. (defaults to '#')
rew
Rewind when this DTMF digit is received. (defaults to '*')
stop
Stop playback when this DTMF digit is received.
pause
Pause playback when this DTMF digit is received.
restart
Restart playback when this DTMF digit is received.
options
o(time):
time - Start at time ms from the beginning of the
file.
[See Also]
Not available
/Johan
Please put some light on these too.
On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer li...@jttech.se
mailto:li...@jttech.se wrote:
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded
file as an audio player. Is it possible that we can use it for
multiple forward and rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys
and implement this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1
minute rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7
and 9 as 15 minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context conference_play_recordings_conference_connect {
playrec_intro = {
Set(position=0);
goto play,1;
}
play = {
while (true) {
if (${position}==-1) { goto recording_end,1; }
//rewind 5 seconds after every action (so the user doesn't
feel lost...)
Set(position=$[${position}-5000]);
if (${position} 0) { Set(position=0); }
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}));
Set(position=${CPLAYBACKOFFSET});
if (${CPLAYBACKSTATUS}==ERROR) {
Playback(pbx_error_500);
Playback(pbx_endcall);
Wait(2);
Hangup();
}
//If stopped by user
if (${CPLAYBACKSTATUS}==USERSTOPPED) {
if (!${ISNULL(${CPLAYBACKSTOPKEY})}) { goto
${CPLAYBACKSTOPKEY},1
Re: [asterisk-users] ControlPlayback's options
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded file as
an audio player. Is it possible that we can use it for multiple
forward and rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys and
implement this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1 minute
rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7 and 9 as 15
minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context conference_play_recordings_conference_connect {
playrec_intro = {
Set(position=0);
goto play,1;
}
play = {
while (true) {
if (${position}==-1) { goto recording_end,1; }
//rewind 5 seconds after every action (so the user doesn't feel
lost...)
Set(position=$[${position}-5000]);
if (${position} 0) { Set(position=0); }
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}));
Set(position=${CPLAYBACKOFFSET});
if (${CPLAYBACKSTATUS}==ERROR) {
Playback(pbx_error_500);
Playback(pbx_endcall);
Wait(2);
Hangup();
}
//If stopped by user
if (${CPLAYBACKSTATUS}==USERSTOPPED) {
if (!${ISNULL(${CPLAYBACKSTOPKEY})}) { goto
${CPLAYBACKSTOPKEY},1; }
}
}
}
recording_end = {
//The end of the recording is reached
Set(position=0);
Background(pbx_endcall);
WaitExten(2);
Hangup();
}
1 = {
Set(position=$[${position}-6]); //Rewind 1 minute
goto play,1;
}
2 = {
//Instructions, that could be aborted with 2.
//1 and 3 could be used to forward/rewind 0 ms effectivly
disabling the defalut..
ControlPlayback(conf_playrec_instructions_full,0,1,3,2);
Wait(1);
goto play,1;
}
3 = {
Set(position=$[${position}+6]); //Forward 1 minute
goto play,1;
}
4 = {
Set(position=$[${position}-30]); //Rewind 5 minutes
goto play,1;
}
5 = {
goto conference_play_recordings_conference_paused, announce, 1;
//Pause
}
6 = {
Set(position=$[${position}+30]); //Forward 5 minutes
goto play,1;
}
7 = {
Set(position=$[${position}-90]); //Rewind 15 minutes
goto play,1;
}
9 = {
Set(position=$[${position}+90]); //Forward 15 minutes
goto play,1;
}
0 = {
goto playrec_intro,1; //Restart playback of the current recording
}
* = {
//Previous recording
//If no recording found, resume playback
goto play,1;
}
# = {
//Next recording
//If no recording found, resume playback
goto play,1;
}
i = {
goto play,1;
}
}
context conference_play_recordings_conference_paused {
announce = {
Set(time=$[${epoch_start}+${position}/1000]);
while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
}
}
i = {
//Every inputs goes here
Wait(1);
goto conference_play_recordings_conference_connect,play,1;
}
}
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
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Med vänlig hälsning
Johan Wilfer email: jo...@jttech.se
JT Tech | Utvecklare webb: http://jttech.se
direkt: +46 31 380 91 01 support: +46 31 380 91 00
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Re: [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing
On 2011-05-03 16:32, Dean Hoover wrote: I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running dahdi show status. I see that sharing IRQs for Digium cards isn't recommended, so I'm trying to set it so each card gets its own. From the few web sites I've read so far, including Digium's FAQ site, I've added ACPI and verified that the BIOS does not give me the ability to manually set the IRQ. I've even taken one of the TE121's out of the server (it isn't being used anyways). Everything I've done so far has not fixed it. All the cards (as well as USB1) all use IRQ 16. The other option given was to use setpci, but I am unfamiliar with that command. I did what I could to try and find the setting (based on what the man page on Ubuntu's web site) where I could see the value 16, but not getting anywhere. I know that this is more of an Asterisk forum than Digium. If I need to put in a case at Digium I will, but wanted to see if there were any suggestions here before I pursued that. Any help would be appreciated. Dean Hoover A month ago I had similar problems with a HP DL360g6 and a HP DL380g7 running Debian 5 Lenny. In the HP DL360g6 I had one TE121. I noticed IRQ misses and the problem was easily reproduced by running dahdi_maint to enable loopback and patlooptest while compiling asterisk to create some i/o. When I installed Debian 6 Squeeze instead the problem went away. Tested with both servers above. On this page I found some information about APIC (Advanced Programmable Interupt Controller) http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html (quite old but informative) I haven't got the time to verify the root cause of the problem yet (I've planned to do this at the end of this month) but my theory is that it has something to do with the kernels APIC handling that was fixed between Debian 5 and 6. Maybe you experience something similar? /Johan -- Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
