September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Implementing h extension in FS
by Ahmed Munir 11 Sep '09

11 Sep '09

Thanks for reply, well actually I'm doing billing after call hangup. If h extension is interupts I'm sending to it to addcdr context which interupts perl script for billing purpose. As I'm listing down below asterisk configuration; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; context add_cdr { _X. => { Hangup(); }; h => { Set(CALL_END_TIME=${EPOCH}); //&print_variables(); NOOP("Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM} Caller-ID:${CALLERID(num)}"); if (${DIALEXECUTED}=YES) { NOOP("Dial-Status:${DIALSTATUS}"); }else { NOOP("Dial was not Executed"); }; DeadAGI(/vopium/agi/billing.pl); NOOP(); }; }; Kindly advice me how I pass/translate h extension in FS in this situation i.e. <action application="api_hangup_hook" data="addcdr 1"/> or there is other way around??? ------------------------------ *From: *Michael Collins <msc(a)freeswitch.org> *Reply-To: *<freeswitch-users(a)lists.freeswitch.org> *Date: *Thu, 10 Sep 2009 00:55:02 -0700 *To: *<freeswitch-users(a)lists.freeswitch.org> *Subject: *Re: [Freeswitch-users] Implementing h extension in FS On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir <ahmedmunir007(a)gmail.com> wrote: HI, I'm newbie in FS, I want to know how to implement h extension of asterisk to FS. As I listed down below; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; My other question is, which application/function/class is use in mod_perl to check the channel status? i.e. busy, answer,hangup,ringing,etc. Kindly advice me soon. -- Regards, Ahmed Munir It depends on what you are trying to accomplish, but the closest thing you'll find in FS to the 'h' extension is the channel variable api_hangup_hook which lets you launch an API at the end of the call. It sounds like you are working on CDR processing. Please tell us more about your application and we'll do our best to offer advice. -MC -- Regards, Ahmed Munir

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Friday Meeting at 11AM CST
by Brian West 11 Sep '09

11 Sep '09

http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Here is the agenda please review and add to it anything you think we should cover. Thanks, Brian

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memory leak
by Benedikt Fraunhofer 11 Sep '09

11 Sep '09

Hello *, a memory leak showed up in our loadtests. It's (still) the same setup as in the http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix. One thing I'd like to add is that "fsctl shutdown restart" was unable to shutdown freeswitch. The last line printed is "switch_core_memory.c:567 Stopping memory pool queue." attached file is a the collected and graphed output of some "ps waux" command. "sz" should be in "physical pages" (2k?). I rerun the test and this time it coredumped trying to malloc() space for some playback. Anything else you need (full backtraces?) to dig into it? Cheers Beni.

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OpenZAP No Audio In Outbound FXO for 8-10 Seconds
by Dan 11 Sep '09

11 Sep '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello all! I am having the following issue. When I dial out over a FXO port (analog) for the first 8-10 seconds I get no audio. If I wait I will eventually hear something. On inbound calls audio works great in both directions. I used ztmonitor to record from the channel and there was in fact audio there. In the recording you can clearly tell when audio starts flowing as you can hear me twice in the recording (I was calling to my cell from a phone on my desk). Any help here would be greatly appreciated! FS is current SVN OpenZAP is current SVN. Hardware: Rhino 8 port w/6 FXO and 2 FXS ports installed. Daniel Morrigan "He who says it cannot be done is interrupting the one doing it." - Chinese Proverb -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqoeHoACgkQ3JaPN6smlEXhdQCfRf5GBKwVrtZWsCS4J1fug2e7 SEEAn1FkyhBPKiXnfUiXgvd2ggqUW87w =OFH/ -----END PGP SIGNATURE-----

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Implementing h extension in FS
by Ahmed Munir 11 Sep '09

11 Sep '09

HI, I'm newbie in FS, I want to know how to implement h extension of asterisk to FS. As I listed down below; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; My other question is, which application/function/class is use in mod_perl to check the channel status? i.e. busy, answer,hangup,ringing,etc. Kindly advice me soon. -- Regards, Ahmed Munir

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REMINDER: Weekly FreeSWITCH Conference Scheduled for Friday, 11AM CST
by Michael Collins 11 Sep '09

11 Sep '09

FYI, We are on schedule for the weekly conference call. Please be sure to join us! More information is available here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Talk to you all tomorrow! -Michael

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Getting core dump from last night's build
by Nik Middleton 10 Sep '09

10 Sep '09

Hi Guys, I'm getting a core dump when running an lua script that's been fine for months In Freeswitch_lua.cpp line 92 is being called, but it's not clear what exactly this is doing lua_State *Session::getLUA() { if (!L) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Doh!\n"); } return L; } Anyone shed a light on this? Regards

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482 Request merged, in serial forking - Solved
by Humberto Quintana 10 Sep '09

10 Sep '09

Hi, I want to share my findings in making work Freeswitch as SBC when Kamailio is doing serial forking. FS doesnt take any routing decision, it receives the R-URI from Kamailio. After the 1st route failed, I was receiving "482 Request merged" for the 2nd route. That was because the SIP transaction "lives" for 4 seconds in FS after receiving the ACK for a 4xx, 5xx, 6xx reply (only when using UDP). Also, Kamailio takes less than 1 second to try a new route for the same call, thus making FS generates the 482 message. To adapt both to my needs. I set in external.xml the timer T4 to 2500 (default is 4000): <settings> .... <param name="timer-T4" value="2500" /> </settings> In Kamailio, I'm executing the sleep function (from cfgutils module) in the failure_route block: failure_route[2] { ...Get your new ruri only for some reply codes... avp_pushto("$ruri", "$(avp(i:1511))"); append_branch(); t_on_failure("2"); #In case of a new failure this block will be executed again sleep("3"); t_relay(); return; } May be it's not the best solution but it could be helpful for someone else. Best regards, Humberto _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404

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FS Crashes After Second Call To Python Application
by Rob Forman 10 Sep '09

10 Sep '09

Hi all, I wrote a small 10-line python script wrapping "txfax" (http://pastebin.freeswitch.org/10274 ). Basically it originates a call with Session(), calls mod_fax txfax application, then hangs up. The weird thing is that this works fine the first I run it from fs_cli. When I run a second time however - it still works and sends the fax - but FS seg faults after the python script exits. Gdb shows something with Session destroy: #0 PYTHON::Session::destroy (this=0x83cd308) at freeswitch_python.cpp: 60 60 Py_DECREF(Self); Here is the backtrace from the core: http://pastebin.freeswitch.org/10275 Is this a bug- or am I not setting up or cleaning up the session properly? I was going to open a jira but thought I'd see if I'm missing something obvious first. Thanks! Rob

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FreeSWITCH configuration
by Carolina Benavides Cabrera 10 Sep '09

10 Sep '09

Hi. I'm looking for any function similar to "rewritehostport" function of OpenSIPS. I need to route the SIP signalling toward Sailfin AS, replacing the RequestURI. Thanks for your help Carolina

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