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Performance
by Krešimir Tonković 17 Jul '08

17 Jul '08

Finally I made the time try Anthony's recommendation for performance enhancement: ------------------------- ulimit -c unlimited 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 244 ulimit -l unlimited comment out the manage-presence option in your sip profile to make the sip a lot less resource intensive ------------------------- (This is quoted from a previous mail on this list). With this, my laptop now takes 2.5 times the original CPS! Thanks Anthony! :-) I notice that my hard drive still trashes heavily while doing high-cps tests, and I suspect that is what holds freeswitch down. Any ideas how to reduce that further? -- Krešimir Tonković

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RTP Bind error
by Krešimir Tonković 17 Jul '08

17 Jul '08

Hi! I'm getting these errors on freeswitch: 2008-07-10 09:30:29 [ERR] sofia_glue.c:1779 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-07-10 09:30:29 [NOTICE] sofia_glue.c:1780 sofia_glue_activate_rtp() Hangup sofia/internal/xxxxxxx(a)192.168.1.10 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Any idea what is going on? -- Krešimir Tonković

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Routing calls based on 'Referred-By'
by Matt Darnell 17 Jul '08

17 Jul '08

Aloha, I receive this snipped packet: INVITE sip:4479139@64.75.xxx.xxx:5060;transport=udp SIP/2.0 To: <sip:4479139@meta.xxx.com> Referred-By: <sip:4473452@meta.xxx.com> Diversion: <sip:4473452@meta.xxx.com>;reason=no-answer I would like the dial plan to route based on the number in the Refered-By or Diversion field. I have 10 DID's from a carrier and when the forward to me, they all have the same number in the invite, they say I should be able to route based on the Diversion field. Is there a way to get the data in the dial plan? Thanks for your help, Matt

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[Fwd: Openser + FreeSwitch Integration.]
by Aadilkhan Maniyar 17 Jul '08

17 Jul '08

Hi All, I would like to try and integrate OpenSER and FS. In this setup OpenSER would be the proxy and the Registrar. The registrations and subscriber info are stored in MySQL database. I would like to use FS as a B2BUA, so that it maintains all the dialog information, different call legs etc. (Something similar to Openser and Asterisk integration) Now my queries: 1. How can I proceed to achieve the above? 2. How can I share the MySQL database so that FS can access the registrations stored by OpenSER. 3. I understand that FS stores its data in SQLite database files. How can I make FS talk to MySQL database. Also I have tried integrating mod_spidermonkey_odbc (http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc) But still the registrations are being stored in the SQLite file and not in the MySQL database. Any help regarding this would be appreciated. Thanks in Advance. Regards, Aadil

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mod_socket_event two-way audio example code?
by tim＠marocon.com 17 Jul '08

17 Jul '08

On the FreeSWITCH wiki page entitled "How does FreeSWITCH compare to Asterisk?" (http://www.freeswitch.org/node/117) there is a pair of sentences that reads "The in-tree mod_event_socket provides a TCP connection on which events can be consumed as well as log data. In addition call control commands may be sent over this interface as well as bi-directional audio flow.". Does anyone have example code that demonstrates this functionality? I've looked through the code and I'm just too new to FreeSWITCH to be able to understand how to approach this. I'll take anything I can get (any language, OS, etc). Thanks, Tim Mattison

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slow hangup detection using FXO into voicemail application
by John Wehle 16 Jul '08

16 Jul '08

> Some FXO devices do not get the hangup signal immediately from the telco. Actually I've done further digging / debugging and it turns out that the FXO device * is * getting the hangup signal immediately from the PBX ... the problem is FreeSWITCH apparently doesn't know how to handle it. It is * shocking * that FreeSWITCH can't handle the basic System 25 PBX :-) hangup signal of ##99 DTMF. Unfortunately that's the only disconnect supervision type supported by the System 25 on vmail lines. Any suggestions on the best place / way to hack FreeSWITCH so that the voicemail application is notified of the hangup when ##99 is received on the line? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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SIP performance tweaking
by Patrick Grondin 16 Jul '08

16 Jul '08

Hi, I am currently doing some characterization testing to figure out the limitations of my freeswitch server. I have tried both with SIPP and by driving SIP calls between 2 Freeswitch through mod_event_socket. I get similar results both ways and I am suspicious of the results I get. Basically, whenever I try to launch calls with SIPP at a rate superior than 8 cps, it seems like calls are being stacked. The response time for a call between the "invite" and the corresponding "200 OK" keeps increasing as long as calls are being made which slows down the processing speed until the average CPS gets around 8 cps. My setup is : 1 Dual-Core AMD Opteron(tm) Processor 8216 @ 2.4 GHz 4 GB of RAM Gigabit NIC Centos 5 FreeSwitch Version 1.0.trunk (8911M) I think that my setup is underperforming and that it might be related to some missing configuration tweaks. I already adjusted the ulimit to the recommended settings in the wiki for maximum performance. I was wondering if there might be something else, that I'm still missing. Would any of you have an idea of what might be holding back my setup ? Thanks ! PG P.S. I gathered a bunch of information that I thought could be useful. Freeswitch performance settings In switch.conf.xml  <param name="max-sessions" value="10000"/>  <param name="sessions-per-second" value="1000"/>  <param name="loglevel" value="alert"/> Here is the SIPP command I used, based on Freeswitch load_test script. # sipp -sf dft_cap.xml 192.168.100.9 -s 9990 -r 15 -rp 1000 -l 200 -d 10000 -i 192.168.100.200 -mi 192.168.100.200 # cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 8216 stepping : 2 cpu MHz : 2411.121 cache size : 1024 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips : 4823.60 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc # cat /proc/meminfo MemTotal: 4042672 kB MemFree: 843188 kB Buffers: 256676 kB Cached: 2671172 kB SwapCached: 1092 kB Active: 777140 kB Inactive: 2192840 kB HighTotal: 0 kB HighFree: 0 kB LowTotal: 4042672 kB LowFree: 843188 kB SwapTotal: 2031608 kB SwapFree: 2016120 kB Dirty: 0 kB Writeback: 0 kB AnonPages: 41068 kB Mapped: 26028 kB Slab: 195536 kB PageTables: 5672 kB NFS_Unstable: 0 kB Bounce: 0 kB CommitLimit: 4052944 kB Committed_AS: 153064 kB VmallocTotal: 34359738367 kB VmallocUsed: 35492 kB VmallocChunk: 34359692283 kB HugePages_Total: 0 HugePages_Free: 0 HugePages_Rsvd: 0 Hugepagesize: 2048 kB # ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 40960 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 40960 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited # top top - 14:31:16 up 1 day, 58 min, 2 users, load average: 5.73, 6.83, 5.36 Tasks: 129 total, 1 running, 128 sleeping, 0 stopped, 0 zombie Cpu(s): 7.0%us, 2.8%sy, 0.0%ni, 50.8%id, 38.8%wa, 0.0%hi, 0.5%si, 0.0%st Mem: 4042672k total, 3418968k used, 623704k free, 256904k buffers Swap: 2031608k total, 15488k used, 2016120k free, 2681820k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 6078 root 18 0 614m 109m 6480 S 19 2.8 3:39.11 freeswitch

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Weekly Conf Call Tomorrow at 9am CST
by Michael Collins 16 Jul '08

16 Jul '08

Here's the new link: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_16 it's pretty bare right now so add some items for discussion tomorrow. I'm sure that the wiki is not perfect! :-) -MC

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uuid_send_dtmf
by Cesar Cepeda 16 Jul '08

16 Jul '08

Would it be posible to add a uuid_send_dtmf API? I coded one copying from another API, it works, but I'm not sure if it's the correct way of doing it (freeing memory, calling functions, etc), anyway I would like it if uuid_send_dtmf is added to the trunk. Here's the code: #define UUID_SEND_DTMF_SYNTAX "<uuid> <dtmf_data>" SWITCH_STANDARD_API(uuid_send_dtmf_function) { switch_core_session_t *psession = NULL; char *mycmd = NULL, *argv[2] = { 0 }; char *uuid = NULL, *dtmf_data = NULL; int argc = 0; if (session) { return SWITCH_STATUS_FALSE; } if (switch_strlen_zero(cmd)) { goto usage; } if (!(mycmd = strdup(cmd))) { goto usage; } if ((argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0])))) < 2) { goto usage; } uuid = argv[0]; dtmf_data = argv[1]; if (switch_strlen_zero(uuid) || switch_strlen_zero(dtmf_data)) { goto usage; } if (!(psession = switch_core_session_locate(uuid))) { stream->write_function(stream, "-ERR Cannot locate session!\n"); return SWITCH_STATUS_SUCCESS; } switch_core_session_send_dtmf_string(psession, (const char *) dtmf_data); goto done; usage: stream->write_function(stream, "-USAGE: %s\n", UUID_SEND_DTMF_SYNTAX); switch_safe_free(mycmd); done: if (psession) { switch_core_session_rwunlock(psession); } switch_safe_free(mycmd); return SWITCH_STATUS_SUCCESS; } And on SWITCH_MODULE_LOAD_FUNCTION(mod_commands_load) I added: SWITCH_ADD_API(commands_api_interface, "uuid_send_dtmf", "send dtmf digits", uuid_send_dtmf_function, UUID_SEND_DTMF_SYNTAX); Thanks. Cesar Cepeda.

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Inbound gateway without registration
by whudson05 16 Jul '08

16 Jul '08

Hi, I've been having difficulties allowing inbound calls from a gateway which isn't authorised. The freeswitch system simply returns a "Proxy authentication required" error. Is there any way which I can set it up so that registration is not required for this inbound gateway? Thanks -- View this message in context: http://www.nabble.com/Inbound-gateway-without-registration-tp18490189p18490… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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