July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to check if a channel variable is present or NULL
by Cavalera Claudio Luigi 01 Jul '09

01 Jul '09

Hello, I'm trying to implement this kind of logic in the dialplan if the channel variable sip_refer_to matches regexp than do action else if sip_refer_to exists (not NULL) but does not match regexp than do anti-action else if sip_refer_to does not exist as a channel_variable (NULL ??) than do another action is there a way to check if a channel variable exists or not? Maybe a special regexp to match? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Freeswitch memory usage is too high
by Muhammad Danish Moosa 01 Jul '09

01 Jul '09

Hi Freeswitch is being used in a scenario where two endpoints are running traffic with bypass media mode. Performance is good and all things are smooth. But as the time goes after starting freeswitch, it starts consuming almost whole of memory. Note , freeswitch is being started with -core option, is it related? If this 99% memory consumption is any red alert, as we can see calls are still connecting fine and all is going as usual. -- Muhammad Danish Moosa

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Any advances on T.38 support for FS?
by François Delawarde 01 Jul '09

01 Jul '09

Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this "modern" version of a 50yo technology to switch to and with FreeSwitch? François.

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Fwd: [UniMRCP] PocketSphinx Plugin Available
by Brian West 01 Jul '09

01 Jul '09

I'm about to wire Arsen the donated money for his work. Remember if you haven't sent me what you have said you would send in please paypal brian(a)freeswitch.org so I can wire it to Arsen. Thanks, Brian Begin forwarded message: > I would like to announce the availability of PocketSphinx ASR plugin > for UniMRCP server. > > PocketSphinx UniMRCP server can be used with an MRCP compliant > client, which supports JSGF grammar. > Currently supported ASR features are as follows: > > Methods: > DEFINE-GRAMMAR > RECOGNIZE > GET-RESULT > START-INPUT-TIMERS > STOP > Events: > START-OF-INPUT > RECOGNITION-COMPLETE > Headers: > Noinput-Timeout > Recognition-Timeout > Completion-Cause > Completion-Reason > Save-Waveform > Grammar: JSGF > > For the instructions on how to build and configure PocketSphinx with > UniMRCP refer to > http://code.google.com/p/unimrcp/wiki/PocketSphinxPlugin > > Please note, everything is working now, nevertheless this is basic > availability only. > I have mostly tested the integrated solution in the following setup > SipPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/PocketSphinx > > However it requires further testing in different environments from > different speakers, e.t.c. > In other words, your feedback is welcome. > > Thanks, > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org

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Javascript session Recording
by Baskar 01 Jul '09

01 Jul '09

*Hi, I have configured outbound call through JavaScript it is working fine but i want the conversation to be recorded . Javascript: sessionA = new Session("{ignore_early_media=true, origination_uuid="+argv[0]+"}sofia/default/sip:"+argv[0]+"@ 192.168.1.135:5066"); sessionB = new Session("sofia/internal/"+ argv[1] +"%192.168.1.77"); rtn = sessionA .recordFile("/tmp/"+ argv[0] +".wav", "", "", 400000000, 500, 3); bridge(sessionA, sessionB); i have 2 legs one is sessionA and sessionB if i record the sessionA leg i get only the sessionA voice recorded. How can i merge both the call (**sessionA and sessionB)** into single file. can any one assist me to resolve this problem. -- Thanks with Regards, N.Baskar *

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How to know my gateway registering is successed??
by Brad Tuan 01 Jul '09

01 Jul '09

As title ,Does FS keep the status of gateways??

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Testing Freeswitch performance led to strange behavior
by Apostolos Pantsiopoulos 01 Jul '09

01 Jul '09

In the process of trying to use Freeswitch in a production environment I conducted a number of performance tests using various servers. It was then that I noticed some strange behavior from FS. When I stripped down the scenario I was using to a simple bridge scenario, I stumbled upon a strange behavior. The scenario as I stated is quite simple. |---------| |---------| | |------- Call from sipp------> | | | sipp | | FS | | | <------ Call back to sipp----| | |---------| |---------| I did not use an RTP stream for my calls just to test the signaling alone. The sipp scenario is the standard uac.xml scenario that can be found integrated to sipp with the following options : Test FS 1: sipp <FS_IP>:5060 -s 55555555 -i <SIPP_IP> -mi <SIPP_IP> -ci <SIPP_IP> -r 10 -d 5000 -l 100 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~(35-60) % (35 during the generation of new calls, 60 during the -l limit imposed by the test) Note : 985 of them had a duration (billsec) of 10 and 15 of them had a duration of 11. I tried raising the call rate and limit... Test FS 1: sipp <FS_IP>:5060 -s 55555555 -i <SIPP_IP> -mi <SIPP_IP> -ci <SIPP_IP> -r 20 -d 5000 -l 200 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~0-30 % (0 during the generation of new calls, 30 during the -l limit imposed by the test) THIS IS WHAT MAKES ME WONDER : The distribution of the durations (billsec - not complete durations) : 183 calls with 10 secs billed duration 110 calls with 11 secs billed duration 238 calls with 12 secs billed duration 447 calls with 13 secs billed duration 22 calls with 14 secs billed duration The sipp scenario is simple "hangup the phone after 10 secs". So, why am I seeing these? Of course that has something to do with the stress the machine has been put through during the second test. But I can see it happening to less stressful conditions (i.e. 15 calls per second) to a smaller extend. I captured one of these calls and verified that when the sipp client hangs up exactly 10 secs after the call start, FS receives the BYE and replies with 200 OK. BUT it does not hang the second leg in a timely manner i.e. it sends a BYE to the sipp server side 1-4 seconds AFTER that. That explains the 11, 12, 13, 14 secs durations seen on the second test. What is more interesting is that I would expect to see in the CDRs the first and second leg to have different durations (since the a leg BYE was received and aknowledged by FS in the correct time) i.e. 10 and 14 secs accordingly. But what I get is the same duration for both legs (14 secs). This in my opinion is very dangerous on production environments (you get charged by your provider more seconds that you charge your clients - or - you falsely charge your clients with bigger durations although they hunged up corectly (and you acknowledged it)). NOTE No 1 : All the performance recommendations found in the wiki has been applied. In fact only the essential modules that could make this scenario work were loaded. NOTE No 2 : I tried using asterisk (as a point of reference - don't get me wrong - I am not trying to start a flame war here). And it succeeded doing on the same hardware 60 calls/sec with a channel limit of 400 sim. calls using only 50% of the cpu (maximum). No under any circumstances I have seen the behavior above (this inability to hang call legs in a timely manner). Even when I pushed asterisk to the limits (80 calls per second 600 max call limit) and it started failing on some calls it never failed to hangup the calls for both legs on exactly 10 secs. NOTE No 3 : As you can tell I was using a very small machine for my tests. When I moved the same tests to larger installations (Quad Core Opterons and Xeons) I got proportional results to the above. NOTE No 4 : The tests were performed in a LAN environment and since there was no RTP involved I think there were no bandwidth issues there. NOTE No 5 : The tests were performed using numerous SVN versions (latest : 13610), the stable version and the 1.0.4pre8 version. NOTE No 6 : Using the -hp switch made no noticeable change in behavior. I am not trying to complain for FS's performance (far from it). I am just somewhat disappointed seeing it performing in such a strange manner when under stress. I would prefer a design that drops the calls after a certain threshold than a design that incorrectly handles them all (I am aware of the max sessions per second in switch.conf.cml - but I am starting to see this behavior even with the cpu idling at about 80%). I don't know if anyone else had the same experience when testing Freeswitch. I can happily supply with all the test details (config files, captures etc) to all interested parties. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Cant register a pointer. What wrong?
by Alexey Lubimov 01 Jul '09

01 Jul '09

I sofia_reg.c:1765 have two user records - good #110 and bad #111. bad.xml: <include> <user id="111"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="$${default_vm_password}"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="111"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Bad"/> <variable name="effective_caller_id_number" value="111"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> good.xml: <include> <user id="110"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="$${default_vm_password}"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="110"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Good"/> <variable name="effective_caller_id_number" value="110"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Good user 110 work without any problem. But user "Bad" user 111 can't register to freeswitch. In log I can see only one message - 2009-06-30 16:31:36.590970 [WARNING] sofia_reg.c:1765 Cant register a pointer. good user exists: freeswitch@internal> user_exists id 110 neolant.ru true and bad user exists! freeswitch@internal> user_exists id 111 neolant.ru true good user don't have attr type: freeswitch@internal> user_data 110(a)neolant.ru attr type -ERR no reply but bad user have attr type! : freeswitch@internal> user_data 111(a)neolant.ru attr type pointer Good user have password: freeswitch@internal> user_data 110(a)neolant.ru param password 123456 But bad user no have param pasword! freeswitch@internal> user_data 111(a)neolant.ru param password -ERR no reply What's wrong in these configuration? How I can debug and resolve these problems?

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Error in OpenZap
by Baskar 01 Jul '09

01 Jul '09

*Hi, **I have installed the latest trunk with A102D Sangoma card when i load the openzap i get this error ** freeswitch(a)localhost.localdomain> load mod_openzap 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' freeswitch(a)localhost.localdomain> 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** How can i resolve the above problem. how can i load the openzap My Configuration files default.xml <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/1/a/${dialed_ext}"/> </condition> </extension> /etc/wanpipe/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 0 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> /usr/local/freeswitch/libs/openzap/conf/openzap.conf [span wanpipe1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 **How can i resolve the above problem. how can i load the openzap. can some one assist me to resolve this process.* * -- Thanks with Regards, N.Baskar *

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