March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Perl script or ESL memory leaks, how to check?
by Dmitry Sytchev 05 Mar '11

05 Mar '11

Hi All, I'm using Net::Server::Prefork togethere with perl ESL to listen for outbound socket connections from Freeswitch in sync mode. Due to use of Prefork, one script instance processes many requests (max_requests). I'm doing database request, finding user to dial and bridge. I noticed that if channel hungs up by initiator, memory consumed by my script is never returned to OS, but if bridge finished by remote side (or freeswitch), memory returns to OS and everything works ok. I thought there are some circular references in my script or something like that, but that would affect both cases. In my case when call is finished by freeswitch by reaching end of script and then end of dialplan, everything works fine, but if call is finished by originating channel, memory leak occurs. How can I trace these leaks in my script or ESL library? What should I start from? -- Best regards, Dmitry Sytchev, IT Engineer

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[Dahdi/FreeTDM] Does FS provide call supervision?
by GillesToo 05 Mar '11

05 Mar '11

Hello, Suprisingly, Asterisk doesn't provide call supervision when used with PCI cards + FXO module + Zaptel/Dahdi. http://www.voip-info.org/wiki/view/Asterisk+func+channel CHANNEL() , the only function that is supposed to provide information, always returns "Up", no matter if Asterisk is still dialing or the callee has actually answered the phone. I assume Freeswitch will have the same issue since FreeTDM relies on Dahdi to work with this type of hardware, but I'd like confirmation of this. Thank you. PS: For those interested, here's the code I naively thought would let Asterisk pause until the callee has gone off-hook: ====== ;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing Offhook, Pre-ring, Unknown exten => start,n,Set(INDEX=0) ;tried "Up", "OffHook", and "Rsrvd", to no avail exten => start,n,While($["${CHANNEL(state)}" != "Up" & ${INDEX} < 10]) exten => start,n,NoOp(Channel still ringing: ${CHANNEL(state)}) exten => start,n,Wait(1) exten => start,n,Set(INDEX=$[${INDEX} + 1]) exten => start,n,EndWhile() ====== -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dahdi-FreeTDM-Does-FS-provide… Sent from the freeswitch-users mailing list archive at Nabble.com.

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route inbound - based on sip account
by Andreas Tuerpe 05 Mar '11

05 Mar '11

Hallo FreeSWITCH Users, I use FS V.0.9.6 on pfSense see on -> [http://doc.pfsense.org/index.php/FreeSWITCH] Symptom: German ISP "Portunity" forward inbound calls without any destination_number. ISP solution tip: FS to register over a second sip account to the provider twice. Any account has a separate number. Based on the channel over which the call come in, I have to decide which number is called. So I need help, which condition assignment must use - howto ??? - which fields can I use? - which syntax I have to use?  thanks in advance tuerpean

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What is wrong with my extension?
by Boris Kovalenko 05 Mar '11

05 Mar '11

Hello! May someone tell me why the 73435 is stripped ??? And if I doing something wrong what is the right way. I'm useing FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300) Here is my extension:  <extension name="top.ctx"> <condition field="${v_tagnet_dst}" expression="^([0-9]+)$"> <action application="log" data="ALERT [top.ctx] - transfer to top.ctx"/> <action application="log" data="ALERT [top.ctx] ORIG: ${v_tagnet_dst} REGEX: $1"/>  <action application="execute_extension" data="ext_translate_src XML features"/> <action application="log" data="ALERT [top.ctx] - transfer to top.ctx $1"/> <action application="transfer" data="$1 XML top.ctx"/> </condition> </extension> Here is the output: 2011-03-04 15:56:42.127837 [ALERT] mod_dptools.c:1174 [top.ctx] - transfer to top.ctx 2011-03-04 15:56:42.129862 [ALERT] mod_dptools.c:1174 [top.ctx] ORIG: 73435230020 REGEX: 230020 -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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mod_cdr_pg_csv
by Cliff Wells 05 Mar '11

05 Mar '11

Is this module still supported? I added it to my modules.conf but during compilation I get the following: Compiling /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c ... cc1: warnings being treated as errors /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'save_cdr': /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:225: warning: ISO C90 forbids mixed declarations and code /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:235: warning: ISO C90 forbids mixed declarations and code /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:248: warning: ISO C90 forbids mixed declarations and code /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: warning: ISO C90 forbids mixed declarations and code make[5]: *** [mod_cdr_pg_csv.lo] Error 1 This seems odd since the Makefile includes modmake.rules which has -std=c99. I guess I could disable -pedantic, but I feel I'm starting down a road that ends somewhere over my head. Regards, Cliff

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Good Colo
by Ali G. Zaidi 04 Mar '11

04 Mar '11

Hi, Can someone recommend me a provider that can rent me physical Linux server for voice traffic. My end-points will be allover the USA. Thanks, AGZ Sent from my iPhone

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Sip Info method
by Sam 04 Mar '11

04 Mar '11

Hello, Can we start recording any conference bridge via sip info method ? It should happen for only that particular bridge users and not all the bridges running on the application. For that, what parameters i should set in the SIP info method to achieve this. Regards Sam

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DTMF not passed through to client
by Johannes Jakob 04 Mar '11

04 Mar '11

Hi, what do I have to do to get RFC2833 DTMF passed to the client in an inbound scenario like this: Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk ? The Upstream is sending the DTMF tones correctly: --- 329 2.114845 UPSTREAMIP FS_IP RTP EVENT 62 Payload type=RTP Event, DTMF Eight 8 --- FreeSWITCH isn't logging much of the dtmf stuff, even with loglevel debug: http://pastebin.freeswitch.org/15541 ... Set 2833 dtmf receive payload to 101 ... 2011-03-03 23:29:59.498466 [ERR] switch_rtp.c:282 Failed DTMF sanity check. ... User's directory entry includes: <param name="dtmf-type" value="rfc2833"/> <param name="pass-rfc2833" value="true"/> I tried adding those in the gateway profile as well, no change. On the asterisk box I don't see those rtpevents in the pcap trace anymore, they just vanish on the FS box. Can somebody help me out here? Or point me to the right wiki page? http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF doesn't really tell me, what the options do and http://wiki.freeswitch.org/wiki/DTMF is something different. Thanks and bye, Johannes

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API command
by Sam 04 Mar '11

04 Mar '11

HI, Suppose if i want to place an external caller in the conference so i should use the below command ? originate freetdm/wp1/a/9322273640 67287006 context &bridge 7050 where 7050 leads to conference . Regds Sam

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Segfault on startup when mod_freetdm is enabled.
by Eric Michel 04 Mar '11

04 Mar '11

I updated to the latest git last night, and started having issues. I assumed I'd screwed up and (not being very adept at unistalling things from source) decided to do a clean install today (including the OS). When I enable mod_freetdm in modules.conf.xml, freeswitch segfaults on startup. If I go back and disable mod_freetdm it starts fine. As this is a clean install the only thing I've changed is modules.conf.xml. Is any one else having this problem? Do I need to file a bug report or am I just missing something really simple? I'm running FreeSWITCH Version 1.0.head(git-64806d2 2011-03-02 18-23-19 +0100). I installed the latest non-beta version of Wanpipe which is 3.5.18, and used "make freetdm" when I compiled it. I've got the Sagnoma A200 card. The OS is CentOS 5.5 64bit, AMD Athalon 7850 Dual-core processor, 2Gb RAM, 2 500Gb HD in a software RAID1. Before I compiled freeswitch I enabled "../../libs/freetdm/mod_freetdm" and "asr_tts/mod_flite" in modules.conf. I've run wancfg_fs. Eric

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