- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Help designing setup
by Jose Suero 24 Jul '13

24 Jul '13

Sorry I see it makes more sense to record CDRs on freeswitch Jose Suero <ms(a)mstn.com> wrote: >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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

1 0

Re: [Freeswitch-users] Help designing setup
by Jose Suero 24 Jul '13

24 Jul '13

Found The OpenSIPS article great. Just one question does anybody knows if I collect CDRs at opensips level how can I adjust calls that have no bye in the event that the freeswitch server fails? "Cal Leeming [Simplicity Media Ltd] " <cal.leeming(a)simplicitymedialtd.co.uk> wrote: >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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

1 0

Help designing setup
by Jose Suero 24 Jul '13

24 Jul '13

Hello all I'm setting up a simple voip service and I need some help, we currently have an opensips server with RTP Proxy on it, that handles registration for all phones, behind it there's one (soon to be several) freeswitch servers that handle sip trunks, voicemail, ivr and so on. Aside for some problems this seems to be working fine. We now have to collect CDR records and limit calls, do prepaid and so on. What do you think would be better: a)Our current setup opensips for registration rtpproxy (or mediaproxy if we use CDRTool) with freeswitch on the back b)Only have opensips as a load balancer and handle registration and RTP with freeswitch, and collect CDRs and limit there c)handle registration at opensips level and RTP with freeswitch (where would we collect CDRs?) thanks in advance Jose Suero

2 1

Accessing Valet_parking variables
by SamyGo 24 Jul '13

24 Jul '13

Dear Users, I've my valet_parking working very well, now I need to know which parking lot my call got parked at in a variable. I need that variable in order to save it in memcache for my SIP proxy to know which FS Server to route the call to for a particular parked call retrieval. Any help would be really appreciated. Thanks, Sammy.

4 12

Apparent performance bottleneck and FS freeze
by Victor Chukalovskiy 23 Jul '13

23 Jul '13

Hello, Need some help finding the bottleneck here. Running 1.2.stable on a decent multi-core vmware platform. No RTP, SIP signaling only. Around 2500 sessions and 30 average CPS load. Essentially, it's a multi-profile .xml config that uses mod_lcr extensively (postgres DB). Yesterday it stopped processing calls with the following log output: 2013-07-21 06:00:37.642678 [CRIT] switch_time.c:967 Virtual Migration Detected! Syncing Clock 2013-07-21 06:01:07.141734 [WARNING] switch_scheduler.c:115 Task was executed late by 19 seconds 1 heartbeat (core) 2013-07-21 06:02:32.402668 [CRIT] switch_time.c:967 Virtual Migration Detected! Syncing Clock 2013-07-21 06:04:42.499744 [WARNING] switch_scheduler.c:115 Task was executed late by 115 seconds 1 heartbeat (core) 2013-07-21 06:04:42.499744 [WARNING] switch_scheduler.c:115 Task was executed late by 74 seconds 2 check_ip (core) 2013-07-21 06:05:15.102683 [CRIT] switch_time.c:967 Virtual Migration Detected! Syncing Clock 2013-07-21 06:05:44.961004 [WARNING] switch_scheduler.c:115 Task was executed late by 22 seconds 1 heartbeat (core) 2013-07-21 06:05:44.961004 [WARNING] switch_scheduler.c:115 Task was executed late by 2 seconds 2 check_ip (core) 2013-07-21 06:05:49.322672 [WARNING] mod_lcr.c:448 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:50.962987 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:50.962987 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:50.962987 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... ................................ 2013-07-21 06:05:53.962672 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:54.422671 [ERR] mod_lcr.c:448 Error connecting 2013-07-21 06:05:54.422671 [WARNING] mod_lcr.c:1779 LCR lookup failed for 14389955082 2013-07-21 06:05:54.962671 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:54.962671 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... ................................ 2013-07-21 06:05:55.962758 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:56.062671 [ERR] sofia_glue.c:6605 Error connecting 2013-07-21 06:05:56.062671 [ERR] sofia_glue.c:6606 Error Opening DB 2013-07-21 06:05:56.062671 [WARNING] sofia_glue.c:6605 Max handles 50 exceeded, blocking.... 2013-07-21 06:05:56.062671 [ERR] sofia_glue.c:6605 Error connecting 2013-07-21 06:05:56.062671 [ERR] sofia_glue.c:6606 Error Opening DB FreeSWITCH stayed unusable since then until it was restarted. FYI core DB is already in the RAM: <param name="core-db-name" value="/dev/shm/core.db" /> Thank you, Victor

4 14

Billing CDR Solution
by Jose Suero 23 Jul '13

23 Jul '13

Can anyone recommend a billing solution for freeswitch to collect CDRs, do post/pre paid and so on? thanks in advance Jose Suero

3 2

suspecting memory leak about mutex
by hkchoi 23 Jul '13

23 Jul '13

Hello freeswitch-users, there is some doubt about memory leak in freeswitch implemetation. when a channel is created, some mutex are created as below. SWITCH_DECLARE(switch_status_t) switch_channel_alloc(switch_channel_t **channel, switch_call_direction_t direction, switch_memory_pool_t *pool) { switch_assert(pool != NULL); if (((*channel) = switch_core_alloc(pool, sizeof(switch_channel_t))) == 0) { return SWITCH_STATUS_MEMERR; } switch_event_create_plain(&(*channel)->variables, SWITCH_EVENT_CHANNEL_DATA); switch_core_hash_init(&(*channel)->private_hash, pool); switch_queue_create(&(*channel)->dtmf_queue, SWITCH_DTMF_LOG_LEN, pool); switch_queue_create(&(*channel)->dtmf_log_queue, SWITCH_DTMF_LOG_LEN, pool); switch_mutex_init(&(*channel)->dtmf_mutex, SWITCH_MUTEX_NESTED, pool); switch_mutex_init(&(*channel)->flag_mutex, SWITCH_MUTEX_NESTED, pool); switch_mutex_init(&(*channel)->state_mutex, SWITCH_MUTEX_NESTED, pool); switch_mutex_init(&(*channel)->thread_mutex, SWITCH_MUTEX_NESTED, pool); switch_mutex_init(&(*channel)->profile_mutex, SWITCH_MUTEX_NESTED, pool); ... } And, It seems that some memory is newly allocated in switch_mutex_init() by comments in the following code. [ switch_apr.h ] /** * Create and initialize a mutex that can be used to synchronize threads. * @param lock the memory address where the newly created mutex will be * stored. * @param flags Or'ed value of: * <PRE> * SWITCH_THREAD_MUTEX_DEFAULT platform-optimal lock behavior. * SWITCH_THREAD_MUTEX_NESTED enable nested (recursive) locks. * SWITCH_THREAD_MUTEX_UNNESTED disable nested locks (non-recursive). * </PRE> * @param pool the pool from which to allocate the mutex. * @warning Be cautious in using SWITCH_THREAD_MUTEX_DEFAULT. While this is the * most optimial mutex based on a given platform's performance charateristics, * it will behave as either a nested or an unnested lock. * */ SWITCH_DECLARE(switch_status_t) switch_mutex_init(switch_mutex_t ** lock, unsigned int flags, switch_memory_pool_t *pool); /** * Destroy the mutex and free the memory associated with the lock. * @param lock the mutex to destroy. */ SWITCH_DECLARE(switch_status_t) switch_mutex_destroy(switch_mutex_t *lock); but, I can't find any code that is related with destroying mutex. In my opinion, altough a memory size is very tiny for creating mutex, the leaking memory will be more bigger as times go on. It doesn't need to destory the mutex when a channel is destory? could someone explain about this? Thank you in advance. Regards, hkchoi

3 2

Speech to text voicemail
by Ashwin Jain 23 Jul '13

23 Jul '13

Hi, I want to use Sphinx for voicemail transcription. I have successfully setup the Sphinx using standard language module hub4.5000.DMP, acoustic module hub4wsj_sc_8k and dictionary cmu07a.dic. I tried this setup but the accuracy was very less. I have read that we need to use custom language module for improving the accurary (as the default one is the universal). It will be great if anyone had worked on this and help me out with this. -- Thanks and Regards, Ashwin Jain

2 2

Compile error with FreeSwitch 1.2.11
by Markus von Arx 23 Jul '13

23 Jul '13

Hi I tried to update to FreeSwitch 1.2.11 but get the following compile error (I'm compiling on Windows with Visual Studio 2012 Update 3): d:\freeswitch\build\freeswitch-1.2.11\src\switch_ivr_async.c(1207): error C2220: warning treated as error - no 'object' file generated [D:\FreeSwitch\Build\freeswitch-1.2.11\w32\Library\FreeSwitchCore.2012.vcxproj] d:\freeswitch\build\freeswitch-1.2.11\src\switch_ivr_async.c(1207): warning C4701: potentially uninitialized local variable 'len' used [D:\FreeSwitch\Build\freeswitch-1.2.11\w32\Library\FreeSwitchCore.2012.vcxproj] Checking the source code of switch_ivr_async.c I think the compiler is correct. In the source code, the lines 1213-1214 that contain the initialization of 'len' should probably be moved to line 1204. Also, the latest version of the file on github ( https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/switch_ivr_async.c) looks exactly like this (fixed with a commit from 3 July 2013: https://github.com/FreeSWITCH/FreeSWITCH/commit/02046950bebf84b6cf56f27321c… ). => What to do now with the 1.2.11 version? Will there be a 1.2.12 that fixes this compiler error? Here are the offending lines of code: 1136: static switch_bool_t record_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type) ... 1143: switch_size_t len; ... 1199: case SWITCH_ABC_TYPE_TAP_NATIVE_WRITE: 1200: { 1201: switch_time_t now = switch_micro_time_now(); 1202: switch_time_t diff; 1203: rh->wready = 1; 1204: 1205: if (!rh->rready) { 1206: unsigned char fill_data[SWITCH_RECOMMENDED_BUFFER_SIZE] = {0}; 1207: switch_size_t fill_len = len; 1208: switch_core_gen_encoded_silence(fill_data, &rh->read_impl, len); 1209: switch_core_file_write(&rh->in_fh, fill_data, &fill_len); 1210: } 1211: 1212: 1213: nframe = switch_core_media_bug_get_native_write_frame(bug); 1214: len = nframe->datalen; ... Regards, Markus

4 5

Calls from api originate disconnect after 5 minutes
by Fraser Redmond 23 Jul '13

23 Jul '13

Some of our calls are getting disconnected after exactly 5 minutes. I've finally narrowed it down to be something to do with how the calls are initiated. We have 2 ways of initiating calls: 1) Direct from a softphone out to a gateway to a landline number - this doesn't disconnect after 5min 2) From our webapp using php and fsock to call a command like this: api originate {otherVarsGetPassedThruHereFromPhp=x}user/12605@$freeswitchDomain 442030112233 (I pass through about 8-10 vars. I've tried removing them all, but it still disconnects.) The call works fine and everything is normal until 5 minutes after the call was answered, at which point it hangs up (usually with a second or two, sometimes up to 10 seconds after the 5 minute mark.) A few things I've already tried and ruled out: - I've stripped out all the javascript files that we run, variables we set, and commands we run, so that the only part of the dialplan that gets executed are just: <action application="bridge" data="sofia/gateway/flowroute/796000#442030112233"/> - Right before it disconnects there are no extra lines in the sip log (other than those to handle the hangup.) There's also no sip messages in the log immediately before it disconnects. - A year ago I set "record_waste_resources=true" and that seemed to fix it at the time - but it may have only fixed the calls direct from the softphone. - In the internal and external profiles I've set rtp-timeout-sec=3600 (it used to be 300, and I thought that might be the cause.) - I've done a file compare between the log outputs of each type of call and theres differences that I'd expect because of the different types of call, but otherwise they're about the same. (Differences in how the codec gets set, and a slightly different route through the dialplan.) A few other details about our setup - I upgrade to the latest version of freeswitch regularly, but the problem has been happening for months or years. - Our production server is on Amazon AWS, so there could be a NAT issue... but it also happens on my dev server on my local network (though that is behind a NAT too, so it could be the NAT between my dev server and the gateway... what would I do about that? And why would it happen with one type of call, but not the other.) - Normally, all our calls are recorded, so I'm not doing bypass-media. The wav file for a 5min recording is just slightly under 10mb, so I had been wondering if that was the limit being hit, but it still disconnects if recording is off. - I've tried with a different gateway and different softphone, and it still happens. - Here's a full log output (with sip) if you want to take a look: http://pastebin.freeswitch.com/21211 http://pastebin.freeswitch.com/21212 (this is direct from the softphone, in case the comparison helps) I'm stumped, so any ideas or advice would be much appreciated. I have a pcap I can send if you want to look at one. My best guess right now is that it's a bug to do with api originate. (That it's setting a variable that a normal call doesn't, or vice-versa.) Or is there something I should change about the format of either: user/12605@$freeswitchDomain or sofia/gateway/flowroute/796000#442030112233 (I set up the format of both of those strings about 3 years ago, so maybe there's a better way to do it now.) Cheers, Fraser

4 3