Re: [asterisk-users] Auto dialer scripts and software
On Friday 17 May 2013, cjwstudios wrote: > A friend asked me for help to auto-dial and play a prerecorded message for > a political campaign. I've briefly googled auto dialer scripts but haven't > seen one that really stands out. Are there any free or cheap auto dial > solutions that you nice folks recommend? You do know that sort of thing is against the law -- or at least requires a permit from the authorities -- in most civilised countries, right? You can get into a *lot* of trouble if you are not careful. If you are quite sure it's legal in your jurisdiction, and you have written permission if required, then it's a simple enough matter just to create a call file that will connect some real-world number with a local extension which just waits for the call to be bridged, then plays a sound file. Easy enough in your favourite scripting language. If the call file is definitely smaller than one block (the size of which depends on your file system), it should be OK to write in situ. Otherwise, write it under /tmp or somewhere and then use the system command "mv" to move it to /var/spool/asterisk/outgoing/ after closing it. -- AJS Answers come *after* questions. -- _ -- 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] Stress testing Asterisk
On 21.05.2013, at 0:05, Tommy Cooper wrote: > Hi, > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > generating are failing. I am trying to run Sipp on the same machine as > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/ has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- 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] Auto dialer scripts and software
On 22/5/13 10:54 am, A J Stiles wrote: You do know that sort of thing is against the law -- or at least requires a permit from the authorities -- in most civilised countries, right? And it's worth adding that even if it is legal in your country, you're almost guaranteed to offend/annoy your target audience. Recorded calls always do. 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
Re: [asterisk-users] Auto dialer scripts and software
Calls on behalf of political candidates are generally legal--even to people on the "do not call" lists. It doesn't seem to be possible to pass legislation preventing them. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Wednesday, May 22, 2013 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Auto dialer scripts and software On 22/5/13 10:54 am, A J Stiles wrote: > You do know that sort of thing is against the law -- or at least > requires a permit from the authorities -- in most civilised countries, right? And it's worth adding that even if it is legal in your country, you're almost guaranteed to offend/annoy your target audience. Recorded calls always do. 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 -- _ -- 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] Fw: Stress testing Asterisk
Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? - Forwarded Message - From: Marie Fischer To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, May 22, 2013 1:16 PM Subject: Re: [asterisk-users] Stress testing Asterisk On 21.05.2013, at 0:05, Tommy Cooper wrote: > Hi, > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > generating are failing. I am trying to run Sipp on the same machine as > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful. -- _ -- 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] Stress testing Asterisk
I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Mitul On Wednesday, May 22, 2013, Tommy Cooper wrote: > Thank you for your help I finally solved this issue. Is it possible that > my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 > core using 3.5 GHz, and 1Gb of RAM? > > - Forwarded Message - > *From:* Marie Fischer 'ma...@vtl.ee');>> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com 'asterisk-users@lists.digium.com');>> > *Sent:* Wednesday, May 22, 2013 1:16 PM > *Subject:* Re: [asterisk-users] Stress testing Asterisk > > > On 21.05.2013, at 0:05, Tommy Cooper 'cvml', 'tomcoope...@yahoo.com');>> > wrote: > > > Hi, > > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > generating are failing. I am trying to run Sipp on the same machine as > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > > Do you have a peer and extension configured for SIPP in your Asterisk > configuration? You also needat least the -s option on > your sipp command line. > > http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. > If the calls still fail, Asterisk console output would be helpful. > > > > -- > _ > -- 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 > > > -- Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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] Changes to the community service maintenance notifications
You may have noticed (or maybe not) that there have been several maintenance notifications for the asterisk.org community services this month. We are working hard to keep up the services running smoothly, and those notices are sent whenever we think our maintenance may interfere with the operation of any of the services. So far, it's been our policy that we send out a maintenance notification whenever we do anything other than the most minor maintenance on the services. You can usually read "may have intermittent availability" as "it should be available unless things go horribly wrong". We now realize that most of these notifications are just spam for most of the community. It is also cumbersome for us to send out the notifications every time we touch the services. Especially considering that the services are typically unavailable for at most a few minutes, if at all. In an effort to reduce spam and make service availability more predictable, we're changing the policy about when we send notifications about community service availability. Starting on Monday, May 27th, we will have a regular maintenance window every Monday for one hour starting at 9:00 PM Central Time (that's 02:00 UTC during daylight saving time in the summer, and 03:00 UTC during standard time). We will try to restrict the service impacting maintenance to that weekly window. For the times where there might be a service interruption outside of that window (either when it needs to be coordinated with our colo provider, or if the maintenance will take longer than one hour), we will send notice of the impending service interruption to just the asterisk-announce mailing list[1]. This will help us in planning service upgrades and maintenance, and reduce the amount of unnecessary email for the community. [1]: http://lists.digium.com/mailman/listinfo/asterisk-announce -- Digium's Asterisk Development Team -- _ -- 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] Fw: Stress testing Asterisk
>From the little experience I have I do not think that that is a good way of >testing the quality of voice. SIP only initiates and eventually terminates the >call, once that the call is connected, SIP and therefore Asterisk are no >longer involved. Once the call is connected it is assigned to a trapsport >layer protocol such as RTP. RTP is the actual protocol that delivers the voice >call between endpoints. I believe that the setup of your network, QoS, codecs >etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani To: Tommy Cooper ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Mitul On Wednesday, May 22, 2013, Tommy Cooper wrote: Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM? > > > >- Forwarded Message - >From: Marie Fischer >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Sent: Wednesday, May 22, 2013 1:16 PM >Subject: Re: [asterisk-users] Stress testing Asterisk > > > >On 21.05.2013, at 0:05, Tommy Cooper wrote: > >> Hi, >> I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is >> generating are failing. I am trying to run Sipp on the same machine as >> Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > >Do you have a peer and extension configured for SIPP in your Asterisk >configuration? You also needat least the -s option on your >sipp command line. >http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. >If the calls still fail, Asterisk console output would be helpful. > > > >-- >_ >-- 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 > > > -- Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422-- _ -- 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] Fw: Stress testing Asterisk
I believe there are options for rtp / audio.. Look at pcap play and rtp echo... Transcoding would be another beast - if you are allowing it Sent from my iPhone 5 On May 22, 2013, at 10:02 AM, Tommy Cooper wrote: > From the little experience I have I do not think that that is a good way of > testing the quality of voice. SIP only initiates and eventually terminates > the call, once that the call is connected, SIP and therefore Asterisk are no > longer involved. Once the call is connected it is assigned to a trapsport > layer protocol such as RTP. RTP is the actual protocol that delivers the > voice call between endpoints. I believe that the setup of your network, QoS, > codecs etc... determine the voice quality of your system. > > > - Forwarded Message - > From: Mitul Limbani > To: Tommy Cooper ; Asterisk Users Mailing List - > Non-Commercial Discussion > Sent: Wednesday, May 22, 2013 3:23 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > I have a question here. > > How can we test the quality of voice upon increasing the call load? > > Can we try passing a voice file using sipp and record the same in dial plan > record application ? Is this reliable enough to simulate near real world > scenario? > > Mitul > > On Wednesday, May 22, 2013, Tommy Cooper wrote: > Thank you for your help I finally solved this issue. Is it possible that my > setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core > using 3.5 GHz, and 1Gb of RAM? > > - Forwarded Message - > From: Marie Fischer > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Sent: Wednesday, May 22, 2013 1:16 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > > On 21.05.2013, at 0:05, Tommy Cooper wrote: > > > Hi, > > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > > generating are failing. I am trying to run Sipp on the same machine as > > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > > Do you have a peer and extension configured for SIPP in your Asterisk > configuration? You also needat least the -s option on > your sipp command line. > http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. > If the calls still fail, Asterisk console output would be helpful. > > > > -- > _ > -- 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 > > > > > -- > Regards, > Mitul Limbani, > Chief Architech & Founder, > Enterux Solutions Pvt. Ltd. > 110 Reena Complex, Opp. Nathani Steel, > Vidyavihar (W), Mumbai - 400 086. India > http://www.enterux.com/ > http://www.entvoice.com/ > email: mi...@enterux.in > DID: +91-22-71967121 > Cell: +91-9820332422 > > > > > -- > _ > -- 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
[asterisk-users] Error 488 Not Acceptable Here
Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- 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] Auto dialer scripts and software
On Wed, 22 May 2013, A J Stiles wrote: If the call file is definitely smaller than one block (the size of which depends on your file system), it should be OK to write in situ. Otherwise, write it under /tmp or somewhere and then use the system command "mv" to move it to /var/spool/asterisk/outgoing/ after closing it. To be an 'atomic' operation, doesn't the 'temporary' directory need to be on the same file system as the 'outgoing' directory? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Fw: Stress testing Asterisk
On 13-05-22 10:02 AM, Tommy Cooper wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani To: Tommy Cooper ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Once upon a time, we set out to create exactly this for testing asterisk. Our goal would have been to run the test every week, comparing the results from the previous week, to make sure asterisk's performance was not getting worse as new commits happened. We came up with the idea of loading testing asterisk using SIPp or some other dialer, then determining at what point asterisk would start failing (performance). We decided the point of failure was quality of audio, since it is usually the first thing to go (even though call control still works). It took a while, but with the help of Leif, we found a tool to analyse audio streams (using MOS score[1]). Basically, you take the original audio file, play it across the network, then record the other side. Then, comparing the two files via Aqua, you get your MOS score. If the score was less then x, you knew asterisk was hitting a performance limit. Track that over time and concurrent calls, you have your metrics. [1] http://www.sevana.fi/aqua.php -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Fw: Stress testing Asterisk
El 22/05/13 12:25, Paul Belanger escribió: On 13-05-22 10:02 AM, Tommy Cooper wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani To: Tommy Cooper ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Once upon a time, we set out to create exactly this for testing asterisk. Our goal would have been to run the test every week, comparing the results from the previous week, to make sure asterisk's performance was not getting worse as new commits happened. We came up with the idea of loading testing asterisk using SIPp or some other dialer, then determining at what point asterisk would start failing (performance). We decided the point of failure was quality of audio, since it is usually the first thing to go (even though call control still works). It took a while, but with the help of Leif, we found a tool to analyse audio streams (using MOS score[1]). Basically, you take the original audio file, play it across the network, then record the other side. Then, comparing the two files via Aqua, you get your MOS score. If the score was less then x, you knew asterisk was hitting a performance limit. Track that over time and concurrent calls, you have your metrics. [1] http://www.sevana.fi/aqua.php Hi! I haven't used it, but there is a quality test algorithm provided by ITU. http://stackoverflow.com/questions/2329403/how-to-start-a-voice-quality-pesq-test http://en.wikipedia.org/wiki/PESQ http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=&arnumber=6043771&queryText%3DDevelopment+of+a+Speech+Quality+Monitoring+Tool+based+on+ITU-T+P.862 - CeSPI Centro Superior para el Procesamiento de la Información Universidad Nacional de La Plata --- Proteja el Medioambiente. No imprima este mail si no es absolutamente necesario -- _ -- 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] Failed to authenticate device "Ext 110"
asterisk users wrote: > > Registration trace > (note that extension 88 is the voicemail extension, which the phone registers > to also for MWI) > --> http://pastebin.com/c3H700wa There are no REGISTER requests in that trace. All I see are SUBSCRIBE, NOTIFY, OPTIONS, and INVITE dialogs. > Call trace: > |Time | 10.8.0.6 | > | | | 192.168.6.2 | > |268.693661| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP > From: "Ext 110" < sip:110@192.168.6.2 To:< sip:88@192.168.6.2 > | |(1024) --> (5060) | > |268.694449| 401 Unauthorized |SIP Status > | |(1024) <-- (5060) | > |268.914195| ACK | |SIP Request > | |(1024) --> (5060) | > |268.945115| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP > From: "Ext 110" < sip:110@192.168.6.2 To:< sip:88@192.168.6.2 > | |(1024) --> (5060) | > |268.945717| 403 Forbidden |SIP Status > | |(1024) <-- (5060) | > |269.041417| ACK | |SIP Request > | |(1024) --> (5060) | This is just a failed INVITE probably due to the username and/or password being incorrect. It's also possible that bad ACLs (see the 'permit/deny/acl' settings in sip.conf) could be to blame. It's hard to say without seeing a full SIP trace and Asterisk CLI output. > I'm also confused by the reference in "sip show peers" to port 5062, as I > can't see that anywhere in the configuration of either the phone or in > sip.conf. All the other phones show port 5060 in the "sip show peers" output. Start there and work through the obvious issues one by one. First, figure out why the phone is showing up on port 5062 and correct it if necessary. Then, double-check the username and password. Keep going down that path until it leads to a resolution or report back to the list if you run into a roadblock. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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 Log rotate not working
Jim, Cron and Logrotate already installed in my machine and already configured as the steps you enlisted. But still logrotate is not running. Date: Tue, 21 May 2013 12:28:31 -0700 > From: Jim Lucas > Subject: Re: [asterisk-users] Asterisk Log rotate not working > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <519bcadf.1000...@cmsws.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 5/21/2013 11:54 AM, Ahmed Munir wrote: > > Checked in /var/logs/ directory, all logs are not rotating by logrotate. > > Please advise how can I overcome this issue as I'm using CentoOS 5 > > Ahmed, > > Proper log rotation depends on a couple things working together > correctly to get the job done. First, you need to make sure you have > the space to rotate the logs. If you have compression enabled, > logrotate creates a copy of the file(s) as it compresses them. You > could be running out of space??? > > Next you need to verify that everything is in place, follow these steps > to do so. Keep in mind that I have CentOS 6.4. So the packages might > differ a little in the name and surely in the version numbering. > > 1) Verify logrotate is installed to your system. > # yum install logrotate > > if it asks you to install it, do so. > > 2) Verify that crond is installed and running. > Below is the output I get when searching yum to see if crond is > installed. If your query returns nothing then crond is not installed. > >[root@jim etc]# yum list all | grep ^cron | grep "@" >cronie.x86_64 1.4.4-7.el6 > @anaconda-CentOS-201303020151.x86_64/6.4 >cronie-anacron.x86_64 1.4.4-7.el6 > @anaconda-CentOS-201303020151.x86_64/6.4 >crontabs.noarch 1.10-33.el6 > @anaconda-CentOS-201303020151.x86_64/6.4 > > If crond is not installed, then you will need to install it. Once > you have it installed, move on to the next step. > > 3) Make sure crond is setup to start at boot time. > >chkconfig crond on > > 4) Verify that logrotate is in one of the cron include folders. Mine > is located in the cron.daily folder. > >[root@jim etc]# find /etc/*/logrotate >/etc/cron.daily/logrotate > >If you don't find that the above file exists, you might need to > re-install logrotate. > > Next I would've had you verify that you have a config file in > /etc/logrotate.d/ for the asterisk log files. But it seems you already > to. After all this, if it still isn't working, double check all the > steps above. > > Let us know if this does or doesn't help. > > -- > Jim Lucas > > > > -- Regards, Ahmed Munir Chohan -- _ -- 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] Stress testing Asterisk
On 22.05.2013, at 16:18, Tommy Cooper wrote: > Thank you for your help I finally solved this issue. Is it possible that my > setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core > using 3.5 GHz, and 1Gb of RAM? Easily, as long as you have no media :) Use -sn uac_pcap instead of -sn uac to test with RTP (and watch your call count drop). Add recording (MixMonitor()) to your dialplan and watch the call count go down even more. ;) A rough way to see if call quality is deteriorating would be to call your Asterisk box while the SIPP test is running and listen to some message played via Background(). > > - Forwarded Message - > From: Marie Fischer > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Sent: Wednesday, May 22, 2013 1:16 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > > On 21.05.2013, at 0:05, Tommy Cooper wrote: > > > Hi, > > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > > generating are failing. I am trying to run Sipp on the same machine as > > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > > Do you have a peer and extension configured for SIPP in your Asterisk > configuration? You also needat least the -s option on > your sipp command line. > http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. > If the calls still fail, Asterisk console output would be helpful. -- _ -- 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] Diversion vs. P-Asserted-Id vs. Remote-Party-Id vs. P-Charge-Info vs. From Fields
We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the lerg... i.e., toll-free... Does anyone have any experience with this? Thanks, Optimistic... -- _ -- 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 Log rotate not working
On Wed, May 22, 2013 at 02:54:46PM -0400, Ahmed Munir wrote: > Jim, > > Cron and Logrotate already installed in my machine and already configured > as the steps you enlisted. But still logrotate is not running. How can you tell that the logrotate cron job was run? At what time it was configured to run? Did you see its output in the logs? And please, do make some minimal effort to RTFM and answer questions on your own. Some tools for your disposal: rpm -ql logrotate | grep cron grep -i crom /var/log/messages Cron jobs which have failed and/or had an output send a message to the user who ran them (root, in your case). Is there a "sendmail" (sendmail, postfix, whatever) running on the system? If so, where does root's mail go to? Read it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
