Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN->TNT->Asterisk->SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP->ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. & memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this:> Run asterisk with "-vvvcg". > Do your test (core file generated). > Run "gdb /usr/sbin/asterisk <core_filename>" > From within gdb run "bt" and send me the output > of it.if it is of use, here it is (from asterisk v.0.5.0) ----------------------------- (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec "longdistance", exten=0x8214488 "H323:8257", priority=2, callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 ----------------------------- If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse
On 15/01/04 19:39, Jesse Peterson wrote:> #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 > #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504Do you experience the same problems when you use the other (bundled) h323 driver? (asterisk/channels/h323/README for instructions) Alastair
Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying "Disconnected from Asterisk server" all of a sudden. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? Thanks ! Tom -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN->TNT->Asterisk->SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP->ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. & memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this:> Run asterisk with "-vvvcg". > Do your test (core file generated). > Run "gdb /usr/sbin/asterisk <core_filename>" > From within gdb run "bt" and send me the output > of it.if it is of use, here it is (from asterisk v.0.5.0) ----------------------------- (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec "longdistance", exten=0x8214488 "H323:8257", priority=2, callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 ----------------------------- If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004
THat's not bad 20 calls through a 800Mhz P3. I new 3Ghz P4 could likely handle 60 then. Not bad. But don't beleive "top". First off if acverages. Think for a minute. We all kow a CPU can never by "20% in use" it is either in an idle loop (at 0%) or doing real work (100%) it can't be in an in-between state. I think with Asterisk what matteris is the probibility that when a packet comes in the CPU is idle and available to process it. When "top" says "20%" that means that is only an 80% chance the CPU is free. Looks like 80 or 85 is about the braking point. Same applies to bandwidth. If a packets needs to go out, it needs to go out NOW not some time later if the sound quality is to be OK. So you look at the "probibility of collision" not just the available bandwidth. If your bandwidth is half used by, say web surfers, then half of your VOIP packets will be delayed. Jitter buffer can help, to a point. Back to CPU utilization: 30% utiliation means it is not available to handle a packet 30% of the time and that hansling must be queued up or delayed. --- Jesse Peterson <jesse@strata-com.com> wrote:> Hello all. I'm new to asterisk and have been using and testing it for > about a week now. My initial hope has been to use it as a sip<->h323 > gateway to tie SIP & H323 based ip phones together with my Cisco > AS5300 and Lucent MaxTNT/MVAM networks. > > I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII > 800 with 256megs RAM. I have tried a couple CVS version from the past > week (maybe 01/09/04 and 01/14/04) and have not been able to get them > to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 > has supported those ok. Primarily test cases have involved sending ip > phone calls via SIP to Asterisk and having Asterisk route the calls > using h323 via a gatekeeper to my TNT network which then sends it out > the PSTN... and the opposite path, PSTN->TNT->Asterisk->SIP Phone. > Another test has been sending a call from a AS5300 using SIP to > Asterisk, out H323 to a TNT. Both of those have worked very well with > the voice quality being excellent (actually better than a SIP->ISDN > T1 hardware solution we've been working with - audiocodes mediant 2k > for those interested). This is the test case I describe below as it > was the one the allowed me to load Asterisk up with the most calls. > > Anyway, I know that what I'm doing is not exactly the intended > primary use of Asterisk. That said, here's what I found. > > Voice quality was very good until I had approx. 25 calls up. At that > point there were intermittent issues with garbled voice, a little > echo, etc. When it reached a little over 30 calls, Asterisk just died > (oops). > During the test, I was trying to keep an eye on proc. & memory util. > Memory never seemed to be an issue - even right before the crash the > Asterisk process was not using more than 20 - 25MB. > Processor utilization was interesting to watch though. I couldn't > make any direct/firm correlation, but it seemed like my spikes were > coming when Asterisk was doing call setup. Even up to about 25 calls, > utilization didn't spike to more the 25% for long, and with ~25 calls > seemed to 'idle' around 15%. Above the 25 (when also started noticing > voice quality issues), the proc. util. seemed to start going wacky - > spikes up to 40, 50, even 60%. Then it went to 99% for a moment, > voice quality was horrible if you could hear anything, and Asterisk > crashed. > > I did not find anything in the logs to inidicate any problems, though > I've found that to be the case pretty much everytime Asterisk > crashes. > > I saw a list thread in which a developer asked for some gdb output... > in it, he said this: > > Run asterisk with "-vvvcg". > > Do your test (core file generated). > > Run "gdb /usr/sbin/asterisk <core_filename>" > > From within gdb run "bt" and send me the output > > of it. > > if it is of use, here it is (from asterisk v.0.5.0) > ----------------------------- > (gdb) bt > #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 > #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at > chan_oh323.c:1504 > #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at > channel.c:1385 > #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, > fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 > #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, > allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at > res_parking.c:224 > #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at > app_dial.c:668 > #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, > data=0x6ef216e8, newstack=1) at pbx.c:396 > #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, > context=0x5de5eeec "longdistance", exten=0x8214488 "H323:8257", > priority=2, > callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", > action=1104606132) at pbx.c:1150 > #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 > #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 > #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 > ----------------------------- > > If anyone has tried something like this or has any comments, I'd be > interested in hearing from them. > > > > jesse > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users====Chris Albertson Home: 310-376-1029 chrisalbertson90278@yahoo.com Cell: 310-990-7550 Office: 310-336-5189 Christopher.J.Albertson@aero.org KG6OMK __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus
I did initially, but I was having problems (possibly just in thinking it through) getting the provided h323 driver to either a) register as a gateway with my gatekeeper - that just does not seem to be and option (please correct me if I'm wrong!!!) or b) setup a 'variable' extension (yes, extensions.conf) that would allow me to route any number to it. jesse -----Original Message----- From: Alastair Maw [mailto:asterisk@almaw.com] Sent: Thu 1/15/2004 5:17 PM To: asterisk-users@lists.digium.com Cc: Subject: Re: [Asterisk-Users] capacity testing On 15/01/04 19:39, Jesse Peterson wrote: > #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 > #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 Do you experience the same problems when you use the other (bundled) h323 driver? (asterisk/channels/h323/README for instructions) Alastair _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4458 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040115/54b74e16/attachment.bin
Sorry for the malformed mail. My responses are marked with '***' below. jesse =====Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying "Disconnected from Asterisk server" all of a sudden. *** The crashes I experienced were fairly transparent. When I had the console (asterisk -r) running, I saw the 'Disconnected' message you mention. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. *** A Xeon of the same speed (800mhz in my case) should certainly perform better - lower, I don't know. I find it a little odd that you experienced basically the opposite of my testing. What version are you running? 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. *** Interesting - I have not experienced that (yet...). 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? *** Since you and I are working in what sounds to be a familiar environment, maybe we should communicate about our test scenarios, etc off list to both help each other and see if we can find some similarities to share with others. Thanks ! Tom -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN->TNT->Asterisk->SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP->ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. & memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this:> Run asterisk with "-vvvcg". > Do your test (core file generated). > Run "gdb /usr/sbin/asterisk <core_filename>" > From within gdb run "bt" and send me the output > of it.if it is of use, here it is (from asterisk v.0.5.0) ----------------------------- (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec "longdistance", exten=0x8214488 "H323:8257", priority=2, callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 ----------------------------- If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. 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Hi, all ! I have a fast question, I am running a few Asterisk systems, but I just noticed one thing quite peculiar. After I started "safe_asterisk", and when I ran PS or TOP, I could see 1 PID "safe_asterisk" and almost 10 PIDs "asterisk -vvvg -c" even when there was no call. However, for the other couple, I started "safe_asterisk" and when I ran PS or TOP, I could see 1 PID "safe_asterisk" and only 1 PID "asterisk -vvvg -c", they are all with Pentium Xeon chip and 512M RAM, no difference in Hardware, and all running the same version of Asterisk on Redhat 7.3. Does anyone have any idea why there is a difference please? The reason that it is important as well is because each "asterisk -vvvg -c" is taking up certain memory and with 10 (more when there are calls) or more of these, I am running into memory problem. However, in the other case, no matter how many calls I have, I only see 1 PID of "asterisk -vvvg -c" and seems that I have less of a memory problem. Any feedback to help solve the mystery? Thanks TOm --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004
Jesse Peterson wrote:>I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. >CVS UPDATE! That code is hardcore old. Jeremy McNamara
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I know, but as I mentioned in the inital post, I haven't been able to get the last 2 cvs versions I've pulled to run stable enough to test. I've seen a 0.7.0 version number mentioned. Is there newer, mostly stable version of code I should try that just hasn't been officially released? jesse -----Original Message----- From: Jeremy McNamara [mailto:jj@nufone.net] Sent: Thu 1/15/2004 10:11 PM To: asterisk-users@lists.digium.com Cc: Subject: Re: [Asterisk-Users] capacity testing Jesse Peterson wrote: >I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. > CVS UPDATE! That code is hardcore old. Jeremy McNamara _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4170 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040115/3ef45efd/attachment.bin
1) Yes, I did get that. I've never seen a segmentation fault message, but that should be b/c I've been running the process in the background since it is obviously seg-faulting. I believe you are also correct that most people are not trying to put the load on it that we are. 2) I always see 'safe_asterisk' and 'asterisk -vvvg' running. my monitoring was always done with top, but I've checked w/ ps a couple times and I believe only ever see 1 of each of those processes. I may have to do some tests again to double check that. My CPU problems did not come until the last 10 - 30 seconds before asterisk crashed. This is still odd that our memory & processor observations are opposite... the next thing I'm going to try is a dual xeon pIII 800 or 1ghz machine to see what happens. 3) I'm running oh323. It was the one I could get to register w/ my gatekeeper as a gateway - that made it much easier for me to do call routing on both sides. I have also noticed some inconsistencies in the call flows like you mention, but haven't taken the time yet to pinpoint exactly what and when they are happening. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of T. Chan Sent: Thursday, January 15, 2004 22:54 To: asterisk-users@lists.digium.com Cc: Alan Chan Subject: RE: [Asterisk-Users] capacity testing Hi all, and Jesse 1. So, you did get the experience of crashing all of a sudden with the "Disconnected from Asterisk server" error message. I got both this and the segmentation error when crashing. I am running the version of asterisk, libpri and zaptel updated to about 5 days ago, but I have had tested Asterisk for more than a month already and needless to say I have had this experience since Day 1, meaning it has always been a problem even in the previous revisions. Henceforth, I feel that it is an intrinsic Asterisk problem, rather than just the problem with specific versions / revisions. I have posted this problem a few times before, I feel that this is a major problem but surprisingly, I was not getting any feedback at all. I have this feeling that more than 90% of the Asterisk community is using the system for PBX application rather than VOIP, may be, just may be, Asterisk has not been tested with a good number of simultaneous calls. 2. I am using Xeon 2.6G chip, much more powerful than yours, I have not got any problem with CPU usage, at least not during the time that I was watching. The thing is when I start 'safe_asterisk' , I could see when doing a PID, 1 "safe_asterisk" PID session and at least 10 (or more especially when there are more calls) "asterisk -vvvg -c" PID session. Each session takes up about 18M to 20M RAM, when that is why I am seeing all very high memory usage. How many sessions of Asterick do you see running after you loaded it? 3. Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's and have had this inbound H323 problem. I tried OH323 (Michael) as well, but for some reasons, I am getting this false connect signal, that is, I made an outbound H323 call to a CiscoAS5300 for example, I heard the ring and immediately on my "Asterisk", it showed call answered when it was still ringing. Do you have that experience?? What setting you have if you do not have that experience? 4. Lets talk off list at utitc@hotmail.com. Thanks Tom -----Original Message----- From: Jesse Peterson [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 8:21 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] capacity testing Sorry for the malformed mail. My responses are marked with '***' below. jesse =====Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying "Disconnected from Asterisk server" all of a sudden. *** The crashes I experienced were fairly transparent. When I had the console (asterisk -r) running, I saw the 'Disconnected' message you mention. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. *** A Xeon of the same speed (800mhz in my case) should certainly perform better - lower, I don't know. I find it a little odd that you experienced basically the opposite of my testing. What version are you running? 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. *** Interesting - I have not experienced that (yet...). 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? *** Since you and I are working in what sounds to be a familiar environment, maybe we should communicate about our test scenarios, etc off list to both help each other and see if we can find some similarities to share with others. Thanks ! Tom -----Original Message----- From: asterisk-users-admin@lists.digium.com [ mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN->TNT->Asterisk->SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP->ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. & memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this:> Run asterisk with "-vvvcg". > Do your test (core file generated). > Run "gdb /usr/sbin/asterisk <core_filename>" > From within gdb run "bt" and send me the output > of it.if it is of use, here it is (from asterisk v.0.5.0) ----------------------------- (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec "longdistance", exten=0x8214488 "H323:8257", priority=2, callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 ----------------------------- If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system ( http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system ( http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/c4e397b0/attachment.htm
On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:> > Does anyone have any idea why there is a difference please? The reason that > it is important as well is because each "asterisk -vvvg -c" is taking up > certain memory and with 10 (more when there are calls) or more of these, I > am running into memory problem.The fact that you see multiple processes is symptomatic of a wonky threading implementation -- all threads should belong to the same process. The exception to this is chan_h323 which actually fork(2)s a new process. Also, as threads share memory space there is no extra usage with extra threads (except for the overhead of keeping the thread state itself, which is minimal). -w -- /~\ The ASCII Ribbon Campaign \ / No HTML/RTF in email X No Word docs in email / \ Respect for open standards
Dear All, So are you saying that I should see 1 PID for "safe_asterisk" and many PIDs for "asterisk -vvvg -c" or just 1 PID for "asterisk -vvvg -c", the problem is I am seeing alot of PIDs for "asterisk -vvvg -c" on a couple of my systems and only 1 PID for "asterisk -vvvg -c" for the other couple, which way is correct and how do I rectify that? When I said seeing multiple PIDs, I mean I actually see like 20 different PID numbers (the second column when I do a ps), not sure if that means 20 different threads or 20 PIDs. How do I take care of wonky threading implementation as you suggested? THanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of asterisk@lists.styx.org Sent: Friday, January 16, 2004 1:33 PM To: asterisk-users@lists.digium.com Cc: Alan Chan Subject: Re: [Asterisk-Users] RE: PID On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:> > Does anyone have any idea why there is a difference please? The reasonthat> it is important as well is because each "asterisk -vvvg -c" is taking up > certain memory and with 10 (more when there are calls) or more of these, I > am running into memory problem.The fact that you see multiple processes is symptomatic of a wonky threading implementation -- all threads should belong to the same process. The exception to this is chan_h323 which actually fork(2)s a new process. Also, as threads share memory space there is no extra usage with extra threads (except for the overhead of keeping the thread state itself, which is minimal). -w -- /~\ The ASCII Ribbon Campaign \ / No HTML/RTF in email X No Word docs in email / \ Respect for open standards _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004