Re: [asterisk-users] Limit Asterisk
Thank you all for the answers. I will do tests to find the problem. One other question I have, in the scenario that I sent, how bad would be to transcode G711 to G729 in 70% of calls? There is a study that shows a statistically loss of performance (concurrent calls) with active transcode? tks 2014-07-24 8:54 GMT-03:00 Scott Griepentrog sgriepent...@digium.com: Whether SSD drives allow you to add any additional calls depends entirely on whether or not they can be written to faster than the SAS drives you have. My experience shows SSD's can be twice as fast as run-of-the-mill SATA, but the performance difference compared to SAS is likely not as great, and could even be worse. You'll need to test two drives to find out. I recommend mounting both to test them and copying a very large ISO file using dd which will give you the transfer rate when finished. Then you should have your answer. On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: 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
Re: [asterisk-users] Limit Asterisk
Another question, what audio format I use in MixMonitor to maintain a connection with reasonable quality and reduce the use of I / O disk? Today I use wav. tks 2014-07-24 9:05 GMT-03:00 Eduardo Leones edua...@ypytecnologia.com.br: Thank you all for the answers. I will do tests to find the problem. One other question I have, in the scenario that I sent, how bad would be to transcode G711 to G729 in 70% of calls? There is a study that shows a statistically loss of performance (concurrent calls) with active transcode? tks 2014-07-24 8:54 GMT-03:00 Scott Griepentrog sgriepent...@digium.com: Whether SSD drives allow you to add any additional calls depends entirely on whether or not they can be written to faster than the SAS drives you have. My experience shows SSD's can be twice as fast as run-of-the-mill SATA, but the performance difference compared to SAS is likely not as great, and could even be worse. You'll need to test two drives to find out. I recommend mounting both to test them and copying a very large ISO file using dd which will give you the transfer rate when finished. Then you should have your answer. On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: 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
[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
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
[asterisk-users] asterisk performace 64bits
Hello, I'm running Asterisk on a CentOS 64-bit server. . Asterisk if I compile using the ./configure --libdir=/usr/lib64 instead of ./configure have a relative gain performace.? Has anyone done any comparison? Is there any way in the compilation or even in settings that I can improve the performace of the asterisk? tks Eduardo -- _ -- 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] Queue wrapuptime and active calls
Hello, My question is this, I have a service queue that members follow the service interval (wrapuptime = 30). However, sometimes these members need to call the customer back, thus making an active call. Occurs when this member disconnects the call shortly following section in the queue already sends a new call for him not understanding the range (wrapuptime = 30). My question is if I can somehow make the queue respect the same range with a call that was not caused by the queue. Is there any way I wrapuptime in a reset after a manual call? tks Eduardo -- _ -- 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] Problem reload queue dynamical members
Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a reload app_queue.so all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem reload queue dynamical members
Josh, thanks for the feedback. That problem can also occur with dynamic members, would not be just for those who work with realtime? tks 2014-06-06 10:14 GMT-03:00 Josh Metzger joshdmetz...@gmail.com: On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones edua...@ypytecnologia.com.br wrote: Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a reload app_queue.so all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? It depends on which exact version of 1.8 you're running, but this appears to be a bug that was fixed in April of this year. From the changelog for 1.8: 2014-04-01 16:48 + [r411584] Joshua Colp jc...@digium.com * apps/app_queue.c: app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of EXITEMPTY. ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ -Josh -- _ -- 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] Application Queue context that calls the extensions
Hello! I wonder what the default context that the Queue application uses to call extensions. If there is a possibility to change this into a context created by me possible? Would you like to get this load value to variables before calling the extension. tks, Eduardo -- _ -- 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] Monitor extension status
Hello, How do I track the status of an extension for socket? I'm trying to use the ExtensionState, but it is returning empty. thank you, Eduardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CPU use
Hello, working in a call center where we set up a structure in asterisk. When my voip reaches 150 calls are with bad quality. We do not transcode codec. What I realized using the top command server (CentOS) processing is too high for the asterisk. But the general processor server is down. Would any limitation of Asterisk to use more hardware resources? tks Eduardo attachment: uso_cpu.PNG-- _ -- 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] Global Variables
I have a question about global variables. Is it possible to somehow keep global variables unset via Dial Plan even Restarting asterisk? tks Eduardo -- _ -- 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] AgentCallBackLogin - stopping
Good afternoon, I have a server still using Asterisk 1.4.44, it has not been done migrating to a more updated. My problem is in AgentCallBackLogin. When many people (more than 100) try to run it at the same time, the application simply hangs or is extremely slow. It takes anywhere from four minutes just to ask for the password of the agent. I thought I might be missing hardware, but both the CPU and Memory are with low consumption. Does anyone have any idea what happens to the AgentCallBackLogin? tks, Eduardo Leones -- _ -- 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] pickup for extension in asterisk 1.4?
Good afternoon,
I'm not able to create a dial plan to pull links extension. I am
using asterisk 1.4.18with the following dial plan:
exten = _ * 7XXX, 1, Pickup ($ {EXTEN: 2} @ PICKUPMARK)
exten = _ * 7XXX, n, Hangup ()
But is not working, the following error appears in the CLI:
- Executing [* 7503 @ ypytrix-05: 2] Pickup
(SIP/501-63a2, 503 @PICKUPMARK) in new stack
[Sep 3 14:54:26] NOTICE [17844]: app_directed_pickup.c: 159 pickup_exec:
Notarget channel found for 503.
- Executing [* 7503 @ ypytrix-05: 3] Hangup (SIP/501-63a2, ) in
newstack
Does anyone know how the pickup for extension in asterisk 1.4?
thanks,
Leones--
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Re: [asterisk-users] pickup for extension in asterisk 1.4?
Already without the spaces, it was time to paste in the email that was well
De: Paul Belanger pabelan...@digium.com
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviadas: Sábado, 3 de Setembro de 2011 15:24
Assunto: Re: [asterisk-users] pickup for extension in asterisk 1.4?
On 11-09-03 02:19 PM, Eduardo Leones wrote:
Good afternoon,
I'm not able to create a dial plan to pull links extension. I am using
asterisk 1.4.18with the following dial plan:
exten = _ * 7XXX, 1, Pickup ($ {EXTEN: 2} @ PICKUPMARK)
exten = _ * 7XXX, n, Hangup ()
remove the spaces to start:
exten = _*7XXX,1,Pickup(${EXTEN:2}@PICKUPMARK)
exten = _*7XXX,n,Hangup()
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] pickup for extension in asterisk 1.4?
friends,
Indeed, following the example of the voip-info, did a SET dial before the
route and it worked perfectly.
example:
exten = 1234.1, Set (__PICKUPMARK = 1234)
exten = 1234, n, Dial (...)
thanks
De: Daniel Tryba dan...@tryba.nl
Para: Asterisk Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviadas: Sábado, 3 de Setembro de 2011 16:56
Assunto: Re: [asterisk-users] pickup for extension in asterisk 1.4?
On Sat, Sep 03, 2011 at 11:19:12AM -0700, Eduardo Leones wrote:
I'm not able?to create?a?dial plan?to pull?links?extension.?I am
using?asterisk?1.4.18with the following?dial plan:
exten =?_?*?7XXX,?1,?Pickup ($ {EXTEN:?2}?@?PICKUPMARK)
exten =?_?*?7XXX, n,?Hangup?()
But?is not working,?the following error appears?in the CLI:
Where are you *setting* the PICKUPMARK?
See the examples on
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
--
Daniel Tryba
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[asterisk-users] Res: Fading voice problem
10% De: Matt Riddell li...@venturevoip.com Para: asterisk-users@lists.digium.com Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28 Assunto: Re: [asterisk-users] Fading voice problem On 3/05/11 10:16 PM, Eduardo Leones wrote: Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from people. Soon I figured some network problem, I did a tcpdump and analyzed by wireshark ...the strange thing is this ... all packets that arrive on the server asterisk are normal or jitter, latency ... But whenAsterisk sends packets to the network or the ISP ... maggoty packages are ... jitter of150ms on average ... latency of more than 1000 ms ... That is, by the way is not the network itself, but the network on the machine ... Dropped iptables to make sure no influence ... I changed the network card and cables... did nothing more ... Anyone have any ideas to help me and chase to find the problem? PS: The server is a CentOS 5.5 - 32 bit ... I've tested the 64bit tb but with the sameerror ... What's your CPU usage like? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Softphone IAX
Anyone know a good IAX2 softphone for Windows that has g729 and it is free? att Eduardo-- _ -- 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] AgentCallbackLogin slow in Asterisk 1.4
Good morning ... I'm using Asterisk 1.4.40 AgentCallbackLogin in a Call Center. What is happening isthat when the Call Center has more than 15 simultaneous calls the login application isextremely slow to fall into the low priority, ie, the agent can log in, but takes about 1minute to drop in priority below. .. I've tried to recompile the asterisk, I installed other version of 1.4, but nothing helped ...Detail that the server is new, very good and even making the conversion to MP3recordings, rarely surpassed 10% for processing. Does anyone have any idea what might be causing this slowdown? I thought of usingthe AddQueueMember, but would have to change much in design, so is my second choice for solution. att Eduardo-- _ -- 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
