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Re: [Freeswitch-users] Capacity testing, seg fault
by Anthony Minessale 30 Nov '07

30 Nov '07

Can you let me into your box while you have it so easy to reproduce so I can examine it and work on it from there? find me on irc... Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "tuhl(a)ix.netcom.com" <tuhl(a)ix.netcom.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, November 29, 2007 5:25:17 PM Subject: Re: [Freeswitch-users] Capacity testing, seg fault I removed the recordFile and replaced it with a 10s sleep, and still got the segfaults. I did see some improvement by going to a 2s sleep instead of a 10s sleep, but then I'm not really testing what I want to be testing - because if the calls are only lasting 2s, that results in much fewer simultaneous channels in use. I will try swapping the roles of the 2 servers. It's not hard. After I do Brian's suggestion and convert away from JS, I'll clean up my scripts and upload them too. Tom At 01:32 PM 11/29/2007, Michael Collins wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C832CF.4FDA2698" Tom, This sounds very interesting. I’d like to know a few things: First, if you disable the call recording on the receiving end, will you still get the segfaults consistently? Just curious to see what the receiver does if it only answers the calls and then hangs up without the extra burden of recording the audio streams. Second, how hard would it be to have the originator and the receiver trade places? If you could, I’d like to see the two machines switch roles, so that the machine currently acting as the receiver makes the calls and the machine acting as the originator will now receive calls. I’m wondering what will happen – will the segfaults stay at the same machine or will they go over to the new receiver, or will they go away altogether…? I know those are kinda brute force suggestions but they might yield some interesting information: If the segfaults occur only on one machine, regardless of whether it’s making or receiving calls then obviously there’s something up with that machine. If the segfaults always occur at the machine receiving the calls then, of course, we’ve got a more interesting issue. Would you mind putting your setup info, scripts, etc. in to the pastebin? Maybe others could try to replicate your symptoms and see what shakes out. Thanks for taking the initiative to do this kind of testing. It will definitely help FS be a better, more stable product. -MC P.S. – I just saw Brian’s emails on this thread, so be sure to check his suggestions as well! From: freeswitch-users-bounces(a)lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of tuhl(a)ix.netcom.com Sent: Thursday, November 29, 2007 12:54 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Capacity testing, seg fault Hi, I'm running some capacity tests on Freeswitch and can cause seg-faults fairly quickly (<1 minute) at a 'light' load of 10 call originations per second. Core dump backtrace is at the bottom, and my debugging shows what looks like corrupted js_session. I'll open an issue on JIRA. I wanted to get opinions on whether this is a valid architecture for testing capacity, and whether I'm making a simple mistake. Environment: I have the trunk version installed on 2 servers and have one server (the originator) calling the other (the receiver) using SIP, g711, with a Gig-E ethernet switch between them. The originating server basically does a session.originate, waitForAnswer, streamFile (10-second 8khz .wav), and hangup. The receiver does a session.answer and a recordFile to a .wav file so I can go back and check voice quality. My capacity testing engine is a Perl script which is using the RPC XML interface to originate the calls on Freeswitch (I submit requests to do a 'jsrun play.js' to a certain phone number, where play.js is a simple script which originates, waitsforanswer, streamfile, hangup) . I can configure it to make a certain number of originations per second and a certain number of total calls. I have no Perl script running on the receiver - I just setup the dialplan to call a .js which answers the call and records it. This testing setup is at an early stage, so right now, to check pass/fail, I just verify that if I ran a 1000-call test on the originator, there should be 1000 .wav files that are all about the same size, on the receiver at the end of the test, and no crashes. I've compiled with debug flags on, and I've set all *_DEBUG flags to 9 (I have also run the tests after a recompile with debug flags off/0, and that didn't make any difference). I've done all the ulimit commands that were in the last few emails on this list. I'm running on FC6 on a Dell 2850 with dual 3.6ghz Xeons (/proc/cpuinfo shows 4 processors), and 4G RAM. I've set max-sessions to 3000 and Session Rate to 100. 'top' is showing freeswitch at 60-80% on the receiver during this test. ISSUE: My problem right now is on the receiver, which I wouldn't care about because I'm most interested in the origination capacity of freeswitch, but with my receiver crashing so quickly, I can't push the originator very hard. I setup my originating engine to make 1000 total calls at 10 call originations per second, each call lasting 10 seconds (which results in about 120 simultaneous channels in use), and I get a seg fault on the receiver within about 500 calls or 50 seconds. If I run a test with 1000 total calls at 6 call originations per second, it will work, but if I run an overnight test with 20,000 total calls at 6 call originations per second, the receiver will sometimes seg-fault at around 15,000 calls, and sometimes it will not. Interestingly though, if I do a very short but very high rate test of 100 total calls at 50 call originations per second, that will usually work. But 500 total calls at 50 calls per second will always seg-fault. Just so you know... I'm shooting for the holy grail of stable operation at 100 call originations per second. I know people have reported much better results than I'm getting. Is something in my setup bad? Here's the core dump backtrace. I added some debug printf's in session_destroy right before the call to destroy_speech_engine, and it looks like the jss has been trampled - for example, jss->flags is always 0 for all my successful calls, but right before it seg-faults, jss->flags is some large random number. This happens every single time. Program terminated with signal 11, Segmentation fault. #0 0x00000000 in ?? () (gdb) bt #0 0x00000000 in ?? () #1 0x40040437 in switch_core_codec_destroy (codec=0x54ece168) at src/switch_core_codec.c:245 #2 0x40ee778b in destroy_speech_engine (jss=0x51206538) at mod_spidermonkey.c:1652 #3 0x40eeaa70 in session_destroy (cx=0x549c9920, obj=0x4eeac7f0) at mod_spidermonkey.c:2723 #4 0x417d1aa7 in js_FinalizeObject (cx=0x549c9920, obj=0x4eeac7f0) at src/jsobj.c:2168 #5 0x417b04d9 in js_GC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1856 #6 0x417af6ad in js_ForceGC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1508 #7 0x417830fd in js_DestroyContext (cx=0x549c9920, gcmode=JS_FORCE_GC) at src/jscntxt.c:285 #8 0x417727ac in JS_DestroyContext (cx=0x549c9920) at src/jsapi.c:956 #9 0x40eec3c9 in js_parse_and_execute (session=0x464d9678, input_code=0x9e31458 "capacity.js", ro=0x0) at mod_spidermonkey.c:3296 #10 0x40eec3f2 in js_dp_function (session=0x464d9678, data=0x9e31458 "capacity.js") at mod_spidermonkey.c:3302 #11 0x40044341 in switch_core_session_exec (session=0x464d9678, application_interface=0x40f26f80, arg=0x9e31458 "capacity.js") at src/switch_core_session.c:936 #12 0x400455be in switch_core_standard_on_execute (session=0x464d9678) at src/switch_core_state_machine.c:169 #13 0x40046605 in switch_core_session_run (session=0x464d9678) at src/switch_core_state_machine.c:406 #14 0x4004381c in switch_core_session_thread (thread=0x9e31288, obj=0x464d9678) at src/switch_core_session.c:681 #15 0x4009047c in dummy_worker (opaque=0x9e31288) at threadproc/unix/thread.c:138 #16 0x007cb3db in start_thread () from /lib/libpthread.so.0 #17 0x0072506e in clone () from /lib/libc.so.6 Tom =============== tuhl(a)ix.netcom.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org =============== tuhl(a)ix.netcom.com ____________________________________________________________________________________ Never miss a thing. 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Re: [Freeswitch-users] Capacity testing, seg fault
by tuhl＠ix.netcom.com 29 Nov '07

29 Nov '07

I removed the recordFile and replaced it with a 10s sleep, and still got the segfaults. I did see some improvement by going to a 2s sleep instead of a 10s sleep, but then I'm not really testing what I want to be testing - because if the calls are only lasting 2s, that results in much fewer simultaneous channels in use. I will try swapping the roles of the 2 servers. It's not hard. After I do Brian's suggestion and convert away from JS, I'll clean up my scripts and upload them too. Tom At 01:32 PM 11/29/2007, Michael Collins wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C832CF.4FDA2698" > >Tom, > >This sounds very interesting. Id like to know a few things: >First, if you disable the call recording on the >receiving end, will you still get the segfaults >consistently? Just curious to see what the >receiver does if it only answers the calls and >then hangs up without the extra burden of recording the audio streams. >Second, how hard would it be to have the >originator and the receiver trade places? If >you could, Id like to see the two machines >switch roles, so that the machine currently >acting as the receiver makes the calls and the >machine acting as the originator will now >receive calls. Im wondering what will happen >will the segfaults stay at the same machine or >will they go over to the new receiver, or will they go away altogether? > >I know those are kinda brute force suggestions >but they might yield some interesting information: >If the segfaults occur only on one machine, >regardless of whether its making or receiving >calls then obviously theres something up with that machine. >If the segfaults always occur at the machine >receiving the calls then, of course, weve got a more interesting issue. > >Would you mind putting your setup info, scripts, >etc. in to the pastebin? Maybe others could try >to replicate your symptoms and see what shakes out. > >Thanks for taking the initiative to do this kind >of testing. It will definitely help FS be a better, more stable product. > >-MC > >P.S. I just saw Brians emails on this thread, >so be sure to check his suggestions as well! > > > >---------- >From: >freeswitch-users-bounces(a)lists.freeswitch.org >[mailto:freeswitch-users-bounces@lists.freeswitch.org] >On Behalf Of tuhl(a)ix.netcom.com >Sent: Thursday, November 29, 2007 12:54 PM >To: freeswitch-users(a)lists.freeswitch.org >Subject: [Freeswitch-users] Capacity testing, seg fault > >Hi, > >I'm running some capacity tests on Freeswitch >and can cause seg-faults fairly quickly (<1 >minute) at a 'light' load of 10 call >originations per second. Core dump backtrace is >at the bottom, and my debugging shows what looks >like corrupted js_session. I'll open an issue on >JIRA. I wanted to get opinions on whether this >is a valid architecture for testing capacity, >and whether I'm making a simple mistake. > >Environment: >I have the trunk version installed on 2 servers >and have one server (the originator) calling the >other (the receiver) using SIP, g711, with a >Gig-E ethernet switch between them. The >originating server basically does a >session.originate, waitForAnswer, streamFile >(10-second 8khz .wav), and hangup. The receiver >does a session.answer and a recordFile to a .wav >file so I can go back and check voice quality. > >My capacity testing engine is a Perl script >which is using the RPC XML interface to >originate the calls on Freeswitch (I submit >requests to do a 'jsrun play.js' to a certain >phone number, where play.js is a simple script >which originates, waitsforanswer, streamfile, >hangup) . I can configure it to make a certain >number of originations per second and a certain >number of total calls. I have no Perl script >running on the receiver - I just setup the >dialplan to call a .js which answers the call >and records it. This testing setup is at an >early stage, so right now, to check pass/fail, I >just verify that if I ran a 1000-call test on >the originator, there should be 1000 .wav files >that are all about the same size, on the >receiver at the end of the test, and no crashes. > >I've compiled with debug flags on, and I've set >all *_DEBUG flags to 9 (I have also run the >tests after a recompile with debug flags off/0, >and that didn't make any difference). I've done >all the ulimit commands that were in the last >few emails on this list. I'm running on FC6 on a >Dell 2850 with dual 3.6ghz Xeons (/proc/cpuinfo >shows 4 processors), and 4G RAM. I've set >max-sessions to 3000 and Session Rate to 100. >'top' is showing freeswitch at 60-80% on the receiver during this test. > >ISSUE: >My problem right now is on the receiver, which I >wouldn't care about because I'm most interested >in the origination capacity of freeswitch, but >with my receiver crashing so quickly, I can't >push the originator very hard. I setup my >originating engine to make 1000 total calls at >10 call originations per second, each call >lasting 10 seconds (which results in about 120 >simultaneous channels in use), and I get a seg >fault on the receiver within about 500 calls or >50 seconds. If I run a test with 1000 total >calls at 6 call originations per second, it will >work, but if I run an overnight test with 20,000 >total calls at 6 call originations per second, >the receiver will sometimes seg-fault at around >15,000 calls, and sometimes it will not. >Interestingly though, if I do a very short but >very high rate test of 100 total calls at 50 >call originations per second, that will usually >work. But 500 total calls at 50 calls per second will always seg-fault. > >Just so you know... I'm shooting for the holy >grail of stable operation at 100 call >originations per second. I know people have >reported much better results than I'm getting. Is something in my setup bad? > >Here's the core dump backtrace. I added some >debug printf's in session_destroy right before >the call to destroy_speech_engine, and it looks >like the jss has been trampled - for example, >jss->flags is always 0 for all my successful >calls, but right before it seg-faults, >jss->flags is some large random number. This happens every single time. > >Program terminated with signal 11, Segmentation fault. >#0 0x00000000 in ?? () >(gdb) bt >#0 0x00000000 in ?? () >#1 0x40040437 in switch_core_codec_destroy >(codec=0x54ece168) at src/switch_core_codec.c:245 >#2 0x40ee778b in destroy_speech_engine >(jss=0x51206538) at mod_spidermonkey.c:1652 >#3 0x40eeaa70 in session_destroy >(cx=0x549c9920, obj=0x4eeac7f0) at mod_spidermonkey.c:2723 >#4 0x417d1aa7 in js_FinalizeObject >(cx=0x549c9920, obj=0x4eeac7f0) at src/jsobj.c:2168 >#5 0x417b04d9 in js_GC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1856 >#6 0x417af6ad in js_ForceGC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1508 >#7 0x417830fd in js_DestroyContext >(cx=0x549c9920, gcmode=JS_FORCE_GC) at src/jscntxt.c:285 >#8 0x417727ac in JS_DestroyContext (cx=0x549c9920) at src/jsapi.c:956 >#9 0x40eec3c9 in js_parse_and_execute >(session=0x464d9678, input_code=0x9e31458 >"capacity.js", ro=0x0) at mod_spidermonkey.c:3296 >#10 0x40eec3f2 in js_dp_function >(session=0x464d9678, data=0x9e31458 "capacity.js") at mod_spidermonkey.c:3302 >#11 0x40044341 in switch_core_session_exec >(session=0x464d9678, >application_interface=0x40f26f80, arg=0x9e31458 "capacity.js") > at src/switch_core_session.c:936 >#12 0x400455be in >switch_core_standard_on_execute >(session=0x464d9678) at src/switch_core_state_machine.c:169 >#13 0x40046605 in switch_core_session_run >(session=0x464d9678) at src/switch_core_state_machine.c:406 >#14 0x4004381c in switch_core_session_thread >(thread=0x9e31288, obj=0x464d9678) at src/switch_core_session.c:681 >#15 0x4009047c in dummy_worker >(opaque=0x9e31288) at threadproc/unix/thread.c:138 >#16 0x007cb3db in start_thread () from /lib/libpthread.so.0 >#17 0x0072506e in clone () from /lib/libc.so.6 > > >Tom > > >=============== >tuhl(a)ix.netcom.com >_______________________________________________ >Freeswitch-users mailing list >Freeswitch-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org =============== tuhl(a)ix.netcom.com

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Capacity testing, seg fault
by tuhl＠ix.netcom.com 29 Nov '07

29 Nov '07

Hi, I'm running some capacity tests on Freeswitch and can cause seg-faults fairly quickly (<1 minute) at a 'light' load of 10 call originations per second. Core dump backtrace is at the bottom, and my debugging shows what looks like corrupted js_session. I'll open an issue on JIRA. I wanted to get opinions on whether this is a valid architecture for testing capacity, and whether I'm making a simple mistake. Environment: I have the trunk version installed on 2 servers and have one server (the originator) calling the other (the receiver) using SIP, g711, with a Gig-E ethernet switch between them. The originating server basically does a session.originate, waitForAnswer, streamFile (10-second 8khz .wav), and hangup. The receiver does a session.answer and a recordFile to a .wav file so I can go back and check voice quality. My capacity testing engine is a Perl script which is using the RPC XML interface to originate the calls on Freeswitch (I submit requests to do a 'jsrun play.js' to a certain phone number, where play.js is a simple script which originates, waitsforanswer, streamfile, hangup) . I can configure it to make a certain number of originations per second and a certain number of total calls. I have no Perl script running on the receiver - I just setup the dialplan to call a .js which answers the call and records it. This testing setup is at an early stage, so right now, to check pass/fail, I just verify that if I ran a 1000-call test on the originator, there should be 1000 .wav files that are all about the same size, on the receiver at the end of the test, and no crashes. I've compiled with debug flags on, and I've set all *_DEBUG flags to 9 (I have also run the tests after a recompile with debug flags off/0, and that didn't make any difference). I've done all the ulimit commands that were in the last few emails on this list. I'm running on FC6 on a Dell 2850 with dual 3.6ghz Xeons (/proc/cpuinfo shows 4 processors), and 4G RAM. I've set max-sessions to 3000 and Session Rate to 100. 'top' is showing freeswitch at 60-80% on the receiver during this test. ISSUE: My problem right now is on the receiver, which I wouldn't care about because I'm most interested in the origination capacity of freeswitch, but with my receiver crashing so quickly, I can't push the originator very hard. I setup my originating engine to make 1000 total calls at 10 call originations per second, each call lasting 10 seconds (which results in about 120 simultaneous channels in use), and I get a seg fault on the receiver within about 500 calls or 50 seconds. If I run a test with 1000 total calls at 6 call originations per second, it will work, but if I run an overnight test with 20,000 total calls at 6 call originations per second, the receiver will sometimes seg-fault at around 15,000 calls, and sometimes it will not. Interestingly though, if I do a very short but very high rate test of 100 total calls at 50 call originations per second, that will usually work. But 500 total calls at 50 calls per second will always seg-fault. Just so you know... I'm shooting for the holy grail of stable operation at 100 call originations per second. I know people have reported much better results than I'm getting. Is something in my setup bad? Here's the core dump backtrace. I added some debug printf's in session_destroy right before the call to destroy_speech_engine, and it looks like the jss has been trampled - for example, jss->flags is always 0 for all my successful calls, but right before it seg-faults, jss->flags is some large random number. This happens every single time. Program terminated with signal 11, Segmentation fault. #0 0x00000000 in ?? () (gdb) bt #0 0x00000000 in ?? () #1 0x40040437 in switch_core_codec_destroy (codec=0x54ece168) at src/switch_core_codec.c:245 #2 0x40ee778b in destroy_speech_engine (jss=0x51206538) at mod_spidermonkey.c:1652 #3 0x40eeaa70 in session_destroy (cx=0x549c9920, obj=0x4eeac7f0) at mod_spidermonkey.c:2723 #4 0x417d1aa7 in js_FinalizeObject (cx=0x549c9920, obj=0x4eeac7f0) at src/jsobj.c:2168 #5 0x417b04d9 in js_GC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1856 #6 0x417af6ad in js_ForceGC (cx=0x549c9920, gcflags=0) at src/jsgc.c:1508 #7 0x417830fd in js_DestroyContext (cx=0x549c9920, gcmode=JS_FORCE_GC) at src/jscntxt.c:285 #8 0x417727ac in JS_DestroyContext (cx=0x549c9920) at src/jsapi.c:956 #9 0x40eec3c9 in js_parse_and_execute (session=0x464d9678, input_code=0x9e31458 "capacity.js", ro=0x0) at mod_spidermonkey.c:3296 #10 0x40eec3f2 in js_dp_function (session=0x464d9678, data=0x9e31458 "capacity.js") at mod_spidermonkey.c:3302 #11 0x40044341 in switch_core_session_exec (session=0x464d9678, application_interface=0x40f26f80, arg=0x9e31458 "capacity.js") at src/switch_core_session.c:936 #12 0x400455be in switch_core_standard_on_execute (session=0x464d9678) at src/switch_core_state_machine.c:169 #13 0x40046605 in switch_core_session_run (session=0x464d9678) at src/switch_core_state_machine.c:406 #14 0x4004381c in switch_core_session_thread (thread=0x9e31288, obj=0x464d9678) at src/switch_core_session.c:681 #15 0x4009047c in dummy_worker (opaque=0x9e31288) at threadproc/unix/thread.c:138 #16 0x007cb3db in start_thread () from /lib/libpthread.so.0 #17 0x0072506e in clone () from /lib/libc.so.6 Tom =============== tuhl(a)ix.netcom.com

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is there any document about how to use analog phone card
by salzh 27 Nov '07

27 Nov '07

Hi, all. I have zma400p and zma800p phone cards. I wanna running it in FreeSWITCH. However, i am not familiar how to configure freeSWITCH to drive the cards. Is there any document about it. Thanks!

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pre-paid calling card app
by Frank - IMPACT 26 Nov '07

26 Nov '07

Has anyone fitted any of the pre-paid calling card apps to FreeSwitch yet? Which ones? I may have a consulting job to implement a pre-paid calling card app on FreeSwitch if there are any out there.

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mod_java - java.lang.NoClassDefFoundError
by jonathan augenstine 26 Nov '07

26 Nov '07

I am encountering the following error when I try to use the mod_java interface: 2007-11-24 18:38:07 [NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute java(fs_ivr arg1 arg2 arg3) Exception in thread "Thread-0" java.lang.NoClassDefFoundError: org/freeswitch/Launcher My dialplan has the following entry: <extension name="Validation_Demo"> <condition field="destination_number" expression="1234"> <action application="java" data="fs_ivr arg1 arg2 arg3"/> </condition> </extension> java.conf.xml has the following entry: <options>  <option value="- Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/> <option value="-Djava.class.path=/usr/local/freeswitch/scripts/fs_ivr"/>  <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address= 127.0.0.1:8000"/> </options> I have also tried setting the class path to freeswitch.jar in the CLASSPATH environment variable. export CLASSPATH=/usr/local/freeswitch/scripts/freeswitch.jar. None of the permutations have met with success. Can anyone spot the error I am making? Any ideas on what to try? Jonathan

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Re: [Freeswitch-users] mem leak and core dump
by Anthony Minessale 25 Nov '07

25 Nov '07

please disable the crash-protection option in switch.conf.xml and execute the command 'ulimit -c unlimited' from your unix shell before you start freeswitch.. here are some other ulimits you may want to turn up for high-volume calls: ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s unlimited ulimit -l unlimited also you can use the unix command "script" to log all the console output to a file so you can capture all of the output for us to examine. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Freeswitch Freeswitch <freeswitch(a)2ride.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, November 23, 2007 10:47:53 AM Subject: Re: [Freeswitch-users] mem leak and core dump Caught signal 11 for unmapped thread! On 11/23/07, Freeswitch Freeswitch <freeswitch(a)2ride.com> wrote: Hello, Now I have updated to the latest trunk as of Nov 21 and moved to a fedora 7 system. Still chasing random crashes of FS. First time this happened last night there were around 1000 sessions up. Crit error was logged referring to a signal caught on an unmapped thread. I wasn't able to catch the specific signal in the console before it scrolled off. This morning I had another crash. This time around 200 sessions running. There was no logging reported during the second crash. Calling scenario is about as simple as it gets. gw1 -> freeswitch -> gw2 all sip codecs, 729 and 711 transcoding off I have been playing with the various rtp knobs in the sofia conf. Sometimes running with rewriting rtp timerstamps, sometimes not. Also tried with soft timer and off. ( can someone point me to an explanation of what the soft timer does? ). Also notable that I compiled the code with debug symbols. No core has been generated during crashes. Is there some way to enable core dumps? Overall I am very impressed with FS - would love to get the cause of these crashes nailed so I can press forward. Any help is very much appreciated. Best regards, Tom On 11/19/07, Michael Jerris < mike(a)jerris.com> wrote: On 11/19/07, Freeswitch Freeswitch <freeswitch(a)2ride.com> wrote: Hello Freeswitch'rs, I recently began using the latest freeswitch beta code as a pure B2BUA for the purpose of topology hiding of IP - PSTN calls. running env: Linux b2bua 2.6.18 #6 SMP Thu Nov 16 10:35:45 EST 2006 i686 Intel(R) Xeon(R) CPU 5140 @ 2.33GHz GNU/Linux Freeswitch version: FreeSwitch Version 1.0.pre1 (6220M) The modules I am using are 711 and 729 codecs, sip, directory, xml_rpc, cdr, xml_cdr, console and syslog. I am seeing what appears to be a memory leak. Consumed memory continues to drop linearly based on number of calls. Also had one core dump. Can you provide details of the call scenarios you are testing, call rates, number of concurrent calls and any other details? What can I do to help troubleshoot these issues? Today I plan to stop using the CDR, rpc and logging modules to see if it makes a difference. We have tested quite heavily with mod_console running, that should not be an issue. Also considering running on one of my fedora core 7 boxes instead of gentoo, again to see if anything changes. We have had repeated issues with gentoo due to patches made either to gcc or to libc or the kernel. It is the only os/distro that I highly suggest never using. Mike _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs

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load average
by Freeswitch Freeswitch 24 Nov '07

24 Nov '07

Does anyone know of kernel tweaks that can be done to optimize running FS on linux? I just completed a test running 800 calls ( 1600 streams ) on a Dell 1950 with dual 2.33GHz and 2 gig of ram. Call scenario was FS running as SIP B2BUA for purposes of topology hiding and dtmf 2833 interworking. Calls were a mix of 711 and 729 Everything worked perfect Load average on the box rose over 50 ( processes in the queue ) at one point. CPU was running between 150% and 180% ( keep in mind there are 2 dual core CPUs so 4 total ) Bandwidth usage was about 80 M Should I worry about that load average number? Seems high but test calls sounded fine. Thanks and best regards, Tom

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Re: [Freeswitch-users] Using a dial-up modem to connect ordinary phones
by Anthony Minessale 24 Nov '07

24 Nov '07

yes mod_openzap supports zaptel compatible hardware as well as pika and sangoma analog and digital boards. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: David Villasmil Govea <david.villasmil(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, November 24, 2007 11:38:35 AM Subject: Re: [Freeswitch-users] Using a dial-up modem to connect ordinary phones Hello, I don´t think you can. But what exactly do you want to do? Dial in and have FS forward the call via a voip service? I don´t know if FS supports an X100. IMO your best bet is buying an X100 clone card, which go for about 15$ on ebay and use asterisk instead of FS. hope this helps On 11/24/07, Ziya Suzen <ziya(a)suzen.net> wrote: > Hi there > > I am quite new to FreeSWITCH and VoIP. I have been searching the net > and the users-list archive, but can't seem to be able to find an > answer to this question: Can I use an old fashion dialup modem to > connect an ordinary phone and hook to VoIP? > > Any pointers, references appreciated. > > Thanks in advance. > > -z > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/

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Re: [Freeswitch-users] Using a dial-up modem to connect ordinary phones
by Anthony Minessale 24 Nov '07

24 Nov '07

Some modems are able to do voice but you can buy inexpensive hardware made specifically for connecting phones and phone lines to voip from a place like PBXEQ http://www.pbxeq.com Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Ziya Suzen <ziya(a)suzen.net> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, November 24, 2007 11:08:45 AM Subject: [Freeswitch-users] Using a dial-up modem to connect ordinary phones Hi there I am quite new to FreeSWITCH and VoIP. I have been searching the net and the users-list archive, but can't seem to be able to find an answer to this question: Can I use an old fashion dialup modem to connect an ordinary phone and hook to VoIP? Any pointers, references appreciated. Thanks in advance. -z _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs

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