February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help debuging core dump
by Nik Middleton 23 Feb '09

23 Feb '09

Hi Guys I'm having problems with seg faults about every 10 mins with call loads > 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards,

2 2

Deployment information and use cases
by Raul Fragoso 23 Feb '09

23 Feb '09

Hello FreeSWITCHERS, My company is currently creating a suite of applications which uses FreeSWITCH as the back-end for an IP-PBX solution. We currently have a prospect to have our first customer installation - a governmental department. That is a tender to have an IP-PBX installation to connect their four office branches, each one with about 300 users - which I am sure FreeSWITCH is able to handle. Since this is an official tender, it's part of their protocol to ask about real sites using the product. Having said that, would you mind sharing some information about your experience with FreeSWITCH deployments ? No need to give many details, but a short summary with company name (if possible), when it was deployed, server equipment, number of users, number of concurrent calls, what kind of functions and services are used and overall capacity of the system. I would really appreciate if you can share that information. And if you guys agree (and explicitly manifest your agreement), I can compile the information in the FreeSWITCH wiki under a "Use Cases" page so it can serve as a common reference as well. Kind regards, Raul Fragoso

7 9

Random problems with cepstral text to speech
by Cavalera Claudio Luigi 23 Feb '09

23 Feb '09

Hello list, sometimes when issue a say command for Cepstral TTS in a conference I get this error: [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking stream handle! [conference_play_file() mod_conference.c:2431] and no audio is played to the conference. I have to restart fs to make it work again, any hint on what it could be? BRs, 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. -----------------------------------------------------------------------------------------------------

3 4

OpenZAP codec Question: Why L16@8000 codec for incoming calls
by Helmut Kuper 23 Feb '09

23 Feb '09

Hello, today I found in FS logfile lines like this: 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms It looks like L16 codec is used for incoming calls: 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal OpenZAP/1:18/2799 [BREAK] 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [proceeding] 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp! 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:18/2799 receive message [TRANSCODING_NECESSARY] 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, State: 0) timed out 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [ready] 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() Remote SDP: v=0^M o=2799 121183017 121183017 IN IP4 85.16.245.254^M s=ATA186 Call^M c=IN IP4 85.16.245.254^M t=0 0^M m=audio 16384 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000/1^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp PCMA/8000 20 ms 160 samples The audio codec compare function finds slightly different codecs for A and B party. The dialplan for incoming calls via openzap is this. I set the codec to use in extensions "bridge" line: <extension name="fp_Local_Extension"> <condition field="destination_number" expression="(491[0-9]|492[0-8])$"> <action application="ring_ready"/> <action application="set" data="ringback=${de-ring}"/> <action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/> <action application="bridge" data="{absolute_codec_string=PCMA}user/$1@$${domain}"/> </condition> </extension> In my vars.xml config I have these codecs configured: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMA"/> So where can I disable the L16 codec, or why is a transcoding necessary? regards Helmut

2 3

Re: [Freeswitch-users] Realm Value
by Ali Al-Rubaie 23 Feb '09

23 Feb '09

I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, Message: 5 Date: Thu, 19 Feb 2009 14:35:29 -0800 From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users(a)lists.freeswitch.org Message-ID: <87f2f3b90902191435p1c9c03aend3303dfb013495b1(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler <intralanman(a)freeswitch.org> wrote: > did you ./configure --enable-core-odbc-suport... those errors reek of > that flag with no unixODBC-devel package installed > > -Ray > Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC **********************

3 2

undefined symbol: krb5_auth_con_getrcache**
by Pekka Kurki 23 Feb '09

23 Feb '09

got this error when starting freeswitch -latest svn 2009-02-23 13:22:50 [CRIT] switch_loadable_module.c:840 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: krb5_auth_con_getrcache** no ccompile errors, krb5libs installed, config with and w/o libcurl. br /pekka

1 0

origination_caller_id_number used instead of effective_caller_id_number
by Alexandru Nedelcu 22 Feb '09

22 Feb '09

I don't know what I'm doing wrong. origination_caller_id_number is used on the B-leg of the bridge, although I also specify the effective_caller_id_number. The thing is that it originally worked in my tests, and I can't figure out what changed in the meantime. The code is something like this ... session = new Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/<phone>"); if (session.ready()) { new_session = new Session("sofia/gateway/provider/<another phone>", session); if (new_session.ready()) bridge(session, new_session); } Thanks,

3 5

recommended settings for max-proceeding param
by Chav Paskov 21 Feb '09

21 Feb '09

Hi Everybody, if it is not too much of a trouble can somebody point to a recommended value for max-proceeding in sofia.conf.xml ? If there is no recommended value what should be taken under consideration in order to determine one. I dug into the archives and discovered a thread called "Freeswitch freezes under increased call load" and there together with session per sec and max allowed sessions was recommended max-proceeding under sofia.conf.xml to be changed. I've just installed 1.0.3 version and checked the sofia.conf.xml file . I was not able to find a default setting for max-proceeding , when i added it i started getting cored umps. Regards Chav

1 0

js session.streamFile() interrupt slowness
by Stephen Crosby 21 Feb '09

21 Feb '09

I have a few scripts that use the javascript session.streamFile('somefile.wav', onDtmf); where onDtmf is a function that returns false to interrupt the streaming file. There is a short delay between the time when I press a key and the time the file stops playing. Is there anything I can adjust that would affect that? It's only maybe 2-3 seconds, but it "feels" too long to me. --Stephen

3 5

Passing Caller_ID to MOD_LCR
by Tchavdar Paskov 21 Feb '09

21 Feb '09

Hi, Is there a way to pass Caller Id to mod lcr and somehow to include it in a custom sql. currently my dialplan looks like this: <extension name="LCR"> <condition field="destination_number" expression="^1(\d+)$" break="never"> <action application="set" data="bypass_media=true" /> <action application="lcr" data="$1"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> i guess that Caller_ID is already passed but i was thinking about making some LCR decisions based on Destination number and Caller_ID /Interstate,Intrastate for example/ Thanks for your time Regards Chav

3 8