[asterisk-users] Asterisk 12.4.0 not able to install pjsip
Hi, I had tried all the steps which I used to inatall Asterisk 12.3.2 Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0 it is not working I am getting XXX in make menuselect resource_module. I tried all trouble shooting steps along with ldconfig etc. I think its a bug can any one help me on this ? -- Regards Sameer Rathod 8109413462 -- _ -- 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 12.4.0 not able to install pjsip
1) What platform are you on (i.e. Ubuntu/Centos/etc) 2) What steps did you take to install the PJSIP libraries? On Wed, Jul 23, 2014 at 7:30 AM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, I had tried all the steps which I used to inatall Asterisk 12.3.2 Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0 it is not working I am getting XXX in make menuselect resource_module. I tried all trouble shooting steps along with ldconfig etc. I think its a bug can any one help me on this ? -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 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
Re: [asterisk-users] Certified Asterisk 11.6 Menuselect
On Mon, Jul 21, 2014 at 9:36 AM, Ryan Wagoner rswago...@gmail.com wrote: Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks for taking note. I've filed an issue here https://issues.asterisk.org/jira/browse/ASTERISK-24104 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
Re: [asterisk-users] Native architecture never available in menuselect
On Mon, Jul 21, 2014 at 12:36 AM, CDR vene...@gmail.com wrote: I want to compile Asterisk always for the native architecture of the machine, and I find that it is never available. It says Depends on: native_arch(E) Can use: N/A Conflicts with: N/A Support Level: core This is Fedora 20 gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1) many thanks Philip I've rarely seen a machine that it isn't available on. The exception for me was a virtualbox machine in one particular case. You may get more help if you describe more about the CPU/architecture that the machine uses. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
Re: [asterisk-users] Asterisk 12.4 IMAP VM Issue - Can't move messages between folders
On 07/17/2014 09:46 AM, Dave Fullerton wrote: Hello all, I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I have asterisk set up to connect to my Dovecot IMAP server and I can leave and retrieve messages from my inbox and old messages. However, I am unable to move messages between folders. I get a message from asterisk stating Sorry the users mailbox can't accept more messages. Here is my setup: In Voicemail.conf I have the following set: imapgreetings=no imapparentfolder=Voicemail imapfolder=Voicemail.Inbox On my imap server, I have the following folder structure: INBOX Sent Junk Drafts Voicemail |-Family |-Inbox |-Work I did a packet capture on my imap server and found that when I go to move a message from Old messages to Family the following happens: Asterisk issues a CREATE Voicemail.Family which succeeds with OK Create completed (The folder is successfully created if it does not exist, I can see it in thunderbird). Then Asterisk issues a COPY 1 Family which fails with NO [TRYCREATE] Mailbox Doesn't exist: Family I don't think Asterisk is using the value of imapparentfolder when copying the message. The COPY command should be COPY 1 Voicemail.Family. Is there something I am missing in my configuration or is this a bug? Thank you -Dave I think I have tracked the issue down to save_to_folder in app_voicemail.c. The third argument to mail_move/mail_copy needs to be different, but my C is not strong enough to know what I need to change it to. Any suggestions? -Dave -- _ -- 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] Return to transferor if no answer after REFER
Hi all, Consider the following scenario. Extension A is in a call with B. At some point, B transfers A to C by sending REFER: A Asterisk B C | | | | |-- INVITE/200 --|-- INVITE/200 --| | : : : : | |--- REFER A r: C | | | | 200 OK -| | | | INVITE C -| | | NOTIFY -| | : : : : If the INVITE to C is answered, B is connected to C, and A is hung up (receives BYE), which is the expected behaviour. However, if C does not answer (is busy, etc), Asterisk hangs up on A and B anyway. What I'd like to have is to connect B back to A instead. Is this possible, and if yes, what am I missing? Any pointers are welcome. Thanks, Jānis -- _ -- 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] Limit Asterisk
people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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] Limit Asterisk
Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 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
[asterisk-users] Any way to get rid of X-Asterisk?
Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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] Any way to get rid of X-Asterisk?
This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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] SIP configuration in realtime static and realtime dynamic
Hi all, I’m currently in the process of familiarizing myself with Asterisk, and am trying to move certain configuration objects (such as SIP peers) into a MySQL database, accessed by Asterisk using the ODBC connector. Now, I’ve imported the sippeers MySQL table from the contrib directory of the Asterisk source, and I could add SIP users in here. However, I currently don’t understand whether this realtime dynamic configuration table is meant to replace or just supplement sip.conf. This is because the sippeers table does not offer certain fields for entries in the [general] section of my sip.conf file, such as the ‚udpbindaddr‘ variable. So, am I supposed to put all that in the database by adding appropriate table columns, or can I leave this in the sip.conf file and chan_sip.so will read both the file and MySQL table once loaded? Also, is there anyway that I could use templates, so that I don’t have to redefine everything for each SIP peer? Thanks a lot for help! Robin signature.asc Description: Message signed with OpenPGP using GPGMail -- _ -- 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] Any way to get rid of X-Asterisk?
Yeah I can do that Anything in sip.conf that we can set? N. On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote: This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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 -- _ -- 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] Limit Asterisk
I would also do some math on the bandwidth requirement. If you divide your disk bandwidth by your recording bit rate what is the theoretical maximum number of calls that you can record at once? Assumes that you have infinite CPU and memory and that you can actually drive the disks at their maximum. If this comes out to 300, you are already there. If it comes out to 3000, you have something wrong in your setup or your assumptions and a target to work towards. What quality are you using in the recording? 44k per second(CD quality sound) uses a lot more bandwidth than 3K (telephone quality) What encoding are you using? How low a bit rate can you use and still have usable recordings? If they are for legal or audit use, you can go pretty low. If you are recording soundtracks for reuse in training or publication, you may require higher bit rates. If you disable recording, how many simultaneous calls can you support? Just to be sure that recording is the issue. Ron On 23/07/2014 4:29 PM, Scott Griepentrog wrote: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br mailto:edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- Digium logo Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Limit Asterisk
Thanks for the feedback. In this case SSD disks you think it solves? Eduardo 2014-07-23 18:01 GMT-03:00 Ron Wheeler rwhee...@artifact-software.com: I would also do some math on the bandwidth requirement. If you divide your disk bandwidth by your recording bit rate what is the theoretical maximum number of calls that you can record at once? Assumes that you have infinite CPU and memory and that you can actually drive the disks at their maximum. If this comes out to 300, you are already there. If it comes out to 3000, you have something wrong in your setup or your assumptions and a target to work towards. What quality are you using in the recording? 44k per second(CD quality sound) uses a lot more bandwidth than 3K (telephone quality) What encoding are you using? How low a bit rate can you use and still have usable recordings? If they are for legal or audit use, you can go pretty low. If you are recording soundtracks for reuse in training or publication, you may require higher bit rates. If you disable recording, how many simultaneous calls can you support? Just to be sure that recording is the issue. Ron On 23/07/2014 4:29 PM, Scott Griepentrog wrote: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Limit Asterisk
Thanks for the feedback. In this case SSD disks you think it solves? 2014-07-23 17:29 GMT-03:00 Scott Griepentrog sgriepent...@digium.com: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 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] Any way to get rid of X-Asterisk?
As far as I know, the code that is set to sent these parameters are static and not affected by the sip.conf settings.. If someone finds otherwise, let me know. On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo sym...@gmail.com wrote: Yeah I can do that Anything in sip.conf that we can set? N. On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote: This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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 -- _ -- 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] Limit Asterisk
Do the calculations for both and see what the answer is. The nice thing about having a model is that you can test configurations without actually having to build one until you are confident that it should work. Ron On 23/07/2014 5:04 PM, Eduardo Leones wrote: Thanks for the feedback. In this case SSD disks you think it solves? 2014-07-23 17:29 GMT-03:00 Scott Griepentrog sgriepent...@digium.com mailto:sgriepent...@digium.com: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br mailto:edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- Digium logo Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Limit Asterisk
Please don't top-post. On Wed, 23 Jul 2014, Eduardo Leones wrote: In this case SSD disks you think it solves? Don't buy hardware until you've identified (either empirical or calculated) the bottleneck. But... SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping out at 460,000 blocks per second. I can remember when 10,000 was big :) -- 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] Limit Asterisk
On 23/7/14 10:29 pm, Steve Edwards wrote: Don't buy hardware until you've identified (either empirical or calculated) the bottleneck. If you've plenty of spare RAM (and at 16GB I'd suggest you probably do), I'd throw in the possibility of recording to RAM disk, then moving the calls to hard disk during your quiet (or closed) hours. SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping out at 460,000 blocks per second. I can remember when 10,000 was big :) This. The 840 is a great bit of kit - we've replaced nearly all our spinning disks with a mix of Samsung 830 and 840 drives. 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] Any way to get rid of X-Asterisk?
From sip.conf.sample in 11.10.0 ;use_q850_reason = no ; Default no ; Set to yes add Reason header and use Reason header if it is available. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Lam Sent: Wednesday, July 23, 2014 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Any way to get rid of X-Asterisk? As far as I know, the code that is set to sent these parameters are static and not affected by the sip.conf settings.. If someone finds otherwise, let me know. On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo sym...@gmail.commailto:sym...@gmail.com wrote: Yeah I can do that Anything in sip.conf that we can set? N. On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.commailto:software...@gmail.com wrote: This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.commailto:sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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 -- _ -- 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] Any way to get rid of X-Asterisk?
From sip.conf.sample in 11.10.0 ;use_q850_reason = no ; Default no ; Set to yes add Reason header and use Reason header if it is available. Using 1.8.7. Shiza Thanks as always guys. N. -- _ -- 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] Any way to get rid of X-Asterisk?
This adds the Q850 reason header but doesn't get rid of the other Asterisk-Hangup headers, at least in v11. On Wed, Jul 23, 2014 at 2:53 PM, Eric Wieling ewiel...@nyigc.com wrote: From sip.conf.sample in 11.10.0 ;use_q850_reason = no ; Default no ; Set to yes add Reason header and use Reason header if it is available. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Lam *Sent:* Wednesday, July 23, 2014 5:07 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Any way to get rid of X-Asterisk? As far as I know, the code that is set to sent these parameters are static and not affected by the sip.conf settings.. If someone finds otherwise, let me know. On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo sym...@gmail.com wrote: Yeah I can do that Anything in sip.conf that we can set? N. On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote: This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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 -- _ -- 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 -- _ -- 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] Limit Asterisk
On Wed, 23 Jul 2014, Chris Bagnall wrote: The 840 is a great bit of kit... The 850 is supposed to be shipping next week. It's got 3d VNAND so the chip geometry can be bigger -- higher speeds and greater reliability. -- 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
