Re: [asterisk-users] Where can get the latest manual our user guide
On 02/08/2013 06:35 AM, Ding Peng wrote: Hi, everybody, Where can I get the manual or user guide of latest asterisk version, 1.11.x? I want to know the syntax and usage of all the supported functions or something like that in the latest version. You can find one on the O'Reilly website. Don't recall the link so you have to google for it. And the Asterisk wiki has a lot of info about version 11. Regards, Patrick -- _ -- 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 performance problem...?
Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following http://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html suggests that the values displayed are no longer representing the computation cost only. However to answer your question: G722 to SLIN16 cost is 9000, reverse direction is 6000 ALAW to SLN16 cost is 17000, reverse direction is 14500 G722 to SLN cost is 9600, reverse direction is 8250 ALAW to SLN cost is 9000, reverse direction is 6000 With regards to the CPU usage per core - inside the VM, where only one core is available, the CPU was close to 100% when the problem started to apear, on the physical server with 4 cores, the cores were evenly loaded at about 30-40%. A single call into the conference consumed between 10-20% depending on whether I have denoise enabled or not. There is no dahdi board installed, I only use the dahdi module for conference timer (note that the problem is also present with the timerfd timing module). BR, Hristo On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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 performance problem...?
Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following http://lists.digium.com/pipermail/asterisk-users/2012-November/276132.html suggests that the values displayed are no longer representing the computation cost only. However to answer your question: G722 to SLIN16 cost is 9000, reverse direction is 6000 ALAW to SLN16 cost is 17000, reverse direction is 14500 G722 to SLN cost is 9600, reverse direction is 8250 ALAW to SLN cost is 9000, reverse direction is 6000 With regards to the CPU usage per core - inside the VM, where only one core is available, the CPU was close to 100% when the problem started to apear, on the physical server with 4 cores, the cores were evenly loaded at about 30-40%. A single call into the conference consumed between 10-20% depending on whether I have denoise enabled or not. There is no dahdi board installed, I only use the dahdi module for conference timer (note that the problem is also present with the timerfd timing module). BR, Hristo On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- 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] SayDigits
Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards -- _ -- 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] SayDigits
HI IS THERE POSIBLE TO MONITOR THE DIGIUM PORTS CHANNEL THROUGH SNMP. IF PASSIBLE MEANS KINDLY SHARE THE SNMP CONFIGURATION OR DOCUMENT FOR THAT. Regards Thangaraj 9994828285 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko Sent: Friday, February 08, 2013 4:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SayDigits Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards -- _ -- 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 --- E-mail Disclaimer : The information contained herein (including any accompanying documents) is confidential and is intended solely for the addressee(s). If you have erroneously received this message, please immediately delete it and notify the sender. Also, if you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or taking any action in reliance on the contents of this message or any accompanying document is strictly prohibited and is unlawful. The organization is not responsible for any damage caused by a virus or alteration of the e-mail by a third party or otherwise. The contents of this message may not necessarily represent the views or policies of Sun Business Solutions Pvt Ltd. --- -- _ -- 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] SayDigits
IS THERE POSIBLE TO MONITOR THE DIGIUM PORTS CHANNEL THROUGH SNMP Please don't hyjack a thread, start a new message. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] SayDigits
Is there a way to slow down or speed up the speed at which SayDigits core show application saydigits [Synopsis] Say Digits. [Description] This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. [Syntax] SayDigits(digits) [Arguments] Not available So, I'd have to say no. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] SayDigits
Am 08.02.2013 13:11, schrieb Doug Lytle: Is there a way to slow down or speed up the speed at which SayDigits core show application saydigits [Synopsis] Say Digits. [Description] This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. [Syntax] SayDigits(digits) [Arguments] Not available So, I'd have to say no. Doug You should write a little AGI-Script instead. -- _ -- 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] SayDigits
On 8/2/13 12:11 pm, Doug Lytle wrote: Is there a way to slow down or speed up the speed at which SayDigits So, I'd have to say no. I suppose potentially you could re-record the sound files to 'say' each digit faster (and with shorter rolloff at the end of each word), then put those into a separate [language] folder in /var/lib/asterisk/sounds, then use those instead in your dialplan. You might even be able to process the existing recordings using your favourite audio editing tool to speed the sound files and reduce the rolloff at the end. No guarantees it'll sound any good, mind. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Google talk not (re)connecting after network down
Hi all, I notice yesterday night while doing tests of uptime that if I unplug my network from the internet, then plug it back, my jabber still shows connection to google, but no outgoing calls are going out, and nothing is coming in (calls are going in google vmail since there is no connection to the pbx) My way of restarting jabber was to kill asterisk and restart it. I'm sure I could have unload then reload the module. Is there any safe feature that can make sure that when Jabber shows CONNECTED , it *IS* actually connected ? Thanks folks ! -- _ -- 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] SayDigits
Hello,
My final solution:
...
same = n,Gosub(dati,s,1(${card}))
[dati]
exten = s,1,NoOp
same = n,Set(say=${LEN(${ARG1})})
same = n,Set(digit=0)
same = n,While($[${digit} ${say}])
same = n,Saydigits(${ARG1:${digit}:1})
same = n,Wait(.75)
same = n,Set(digit=$[${digit} + 1])
same = n,Endwhile
same = n,Return
Thank you for yours suggestion
regards
--
_
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Re: [asterisk-users] asterisk 1.8.10.1 meetme
Hello Jonathan, I thank you for prompt reply to my post. I'm using SIP trunks with Polycom sp450 devices. Also, I was wrong to mention meetme, my conference does not involve using meetme feature on Asterisk. It does not happen often, it happens random. On Thu, Feb 7, 2013 at 3:16 PM, Jonathan Rose jr...@digium.com wrote: motty cruz wrote: Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem? Let me just start by saying that MeetMe has been touched by a rather large number of patches in the 11 months and it's quite likely that your problem will be fixed if you upgrade. r373242 comes to mind in particular. Other than that though, it would be helpful if you added some additional information, such as what arguments are are running meetme with and what kinds of devices you are connecting with (SIP phones presumably?) -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Somewhat OT: Specific SIP packets can cause ethernet controller reset
Update with a response to the statement from Intel: http://blog.krisk.org/2013/02/packets-of-death-update.html On Wed, Feb 6, 2013 at 11:08 AM, Kristian Kielhofner k...@kriskinc.com wrote: While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner -- Kristian Kielhofner -- _ -- 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] access control softphone registration through asterisk
Hi, I wana control my SIP register from asterisk. I other hand, when users login into their softphone, dont access to call and when I give them access, they can call. I dont know it's right way to plan my scenario/? -- _ -- 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] RPM updates
On Monday, January 28, 2013 08:06:38 AM Anthony Messina wrote: On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. I'm not sure which CentOs you're using, but I' build them for CentOS/EL 6: See http://messinet.com/rpms/ Of course, if you're looking for the latest possible build, it might take me a few days: https://admin.fedoraproject.org/updates/rpm-4.10.2-2.fc18 As a side note, I've been working out how to move forward with kernel module signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for signed kernel modules are looking good. Though I wish Digium would just get DAHDI into the upstream kernel already :/ -A As of the update to rpm: https://admin.fedoraproject.org/updates/FEDORA-2013-2107, I'm now able to build EL6 packages again. I should have builds for dahdi-linux and dahdi- tools in the repos within an hour or so. (http://messinet.com/rpms). Also, for Fedora 18, and those interested in testing UEFI/Secure Boot and third-party kernel module signing, I've been working out the signed kernel module buildsystem integration thing and will post my public kernel module signing key to http://messinet.com/rpms sometime tonight. Fedora 18 DAHDI- Linux versions greater than dahdi-linux-2.6.2-0.2.rc1 will have the kernel modules signed. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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
