March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sip redirect contact variable no more available in SVN 12638
by rod 17 Mar '09

17 Mar '09

Hi, running SVN r12638, I don't have access anymore to these 2 variables after a SIP 302 message, using info application: variable_sip_redirect_contact_user_0 variable_sip_redirect_contact_host_0 It was okay with SVN r12611. regards, rod

1 0

Re: [Freeswitch-users] Possible memory / cpu leak
by Chris Fowler 17 Mar '09

17 Mar '09

>> Brian: Can you update to SVN trunk as of now? I updated - version reports: FreeSWITCH Version 1.0.trunk (12631) Only difference I note with this build is that upon "shutdown" FS now SegFaults. The mem/cpu usage continues to slowly climb. <snip> 2009-03-16 20:59:32 [CONSOLE] switch_loadable_module.c:1237 do_shutdown() Stopping: mod_spidermonkey Segmentation fault (core dumped) >> Anthony - nothing special is a bit vague. I modified the dial plan to accept extension in the 1000-1029 range Added DialByLast name using directory.lua (from the wiki) Modified the ivr config to play company specific greetings Voicemail is used a few times per day freeswitch@ip-10-250-155-18> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@xxx.xxx.xxx.xx:5060 RUNNING (0) external profile sip:mod_sofia@xxx.xxx.xxx.xx:5080 RUNNING (0) sip.flowroute.com gateway sip:xxxxxx@sip.flowroute.com REGED inphonex gateway sip:xxxxx@sip.inphonex.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx@east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx@east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx@east.callwithus.com REGED default alias internal ALIASED nat alias external ALIASED xxxxx.bbbb.aaa alias internal ALIASED outbound alias external ALIASED ================================================================================================= >> Anthony - valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg Nothing got logged, here's the output. Did I invoke valgrind incorrectly? ==32545== Memcheck, a memory error detector. ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==32545== Using LibVEX rev 1658, a library for dynamic binary translation. ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==32545== For more details, rerun with: -v ==32545== ==32545== My PID = 32545, parent PID = 32511. Prog and args are: ==32545== /usr/local/freeswitch/bin/freeswitch ==32545== -vg ==32545==

2 1

Don't forget about FreeSWITCH-biz mailing list.
by Brian West 17 Mar '09

17 Mar '09

http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz If you want to do business related posts... ads... discuss services and such that don't belong on the users or dev lists please join the freeswitch-dev list. Thanks, Brian West

3 3

Re: [Freeswitch-users] FS DB
by Ali Al-Rubaie 17 Mar '09

17 Mar '09

Actually, I'm trying to explore what data is stored in the DB and how can we retrieve it, if it is posible. You may write some scripts to retrieve the data and use it in other supporting application. Message: 7 Date: Mon, 16 Mar 2009 12:24:34 -0400 From: Michael Jerris <mike(a)jerris.com> Subject: Re: [Freeswitch-users] FS Database To: freeswitch-users(a)lists.freeswitch.org Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0(a)jerris.com> Content-Type: text/plain; charset="us-ascii" Which data? On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > Hi, > > Is it possible to access FS DB to retrieve data? Where can i find > details about that?

2 1

Re: [Freeswitch-users] Possible memory / cpu leak
by Chris Fowler 16 Mar '09

16 Mar '09

>> Jay : what happens in your dialplan ? Nothing special; no external script execution just default pattern matching to route to extensions (per the stock config). >> Brian: Can you update to SVN trunk as of now? Yup, I will pull the trunk and report back in 24 hours. Chris.

3 2

Possible memory / cpu leak
by Chris Fowler 16 Mar '09

16 Mar '09

Hi, Ive been seeing an issue where FreeSWITCHs CPU and memory utilization climb over time; a restart of FS clears up the problem. See graphs for the past week. http://cfowl.postinbox.com/fs.jpg Observed on the Release Candidate, and then upgraded to the current trunk a couple of times. Currently running version FreeSWITCH Version 1.0.trunk (12604). This is seen both when FS is being used (~200 calls/day, and over the weekend when ~5 calls/day). How can I best debug this? Thanks, Chris.

3 2

Outbound Codec
by Ali Al-Rubaie 16 Mar '09

16 Mar '09

Hi, Is there a way to configure FS to offer a specific codec for B regardless of the codec chosen for A leg? So, when FS invites B, it will offer only one codec specified by the administrator. Thanks,

3 2

sip trunking question
by Steven Ward 16 Mar '09

16 Mar '09

I'm trying to set up a sip trunk between a FS and * box, and right now I'm having trouble getting things set up so I make a call from a sip phone registered with my FS box to a sip phone registered w/ my Asterisk box. I have a gateway defined as in directory/default/example.com.xml and in my dialplan I'm trying to do a bridge w/ something like "sofia/gateway/${default_gateway}/12345." When I try to make the call I see from the console: ... New Channel sofia/external/12345 ... ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] The Originate fails. I tried sticking to what the instructions laid out for this in the Connecting FS and Asterisk wiki page, so I'd appreciate some help in figuring out what's going on. Thanks.

4 11

Module Mod Native File - How to handle multiple rates under the same codec.
by Keith Laaks 16 Mar '09

16 Mar '09

Hi, To minimize/eliminate transcoding, I am using mod native file, with a set of transcoded prompts with the appropriate set of file extensions. Everything works as advertised when using the traditional codecs such as pcma, g729, gsm. But speex support is a bit of a challenge. Freeswitch supports: speex@8000h@20i, speex@16000h@20i, speex@32000h@20i. When I setup calls using these various codec flavors, I can see using info (and of course detect by ear) that indeed the call is running at the different 'rates'. I see that regardless of which one I use, the "variable_read_codec" and "variable_write_codec" remains "SPEEX", but depending on the flavor, a "variable_read_rate" and "variable_write_rate" of either 8000,16000 or 32000. But when I try play a file when in 16000 or 32000, I get: 2009-03-15 16:31:13 [INFO] mod_native_file.c:81 native_file_file_open() Opening File [/usr/local/freeswitch/sounds/en/us/callie/all/16000/SUCCESS.SPEEX] 8000hz 2009-03-15 16:31:13 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match. I created my SPEEX files using: speexenc -w <infilename> <outfilename> (note the -w option for 16kHz wideband) So, even though the call is setup using a wideband 16kHz codec, it appears that mod native file is expecting a 8kHz file for all the SPEEX flavors. What am I missing here? Is this module limited to 8KHz rates? I am on 1.0.trunk (12530M). I have not yet looked at these codecs: G7221@16000h, G7221@32000h, CELT@32000h, CELT@48000h, but as these also have multiple rates for the same codec - I expect same issue. I am using ${ variable_read_rate } in the filename path, so fs looks at a set of files encoded with a matching sample rate. But looks like it's always looking for a 8KHz file. If you have had any experience with this, please let me have your advice. Thanks Keith

1 0

FS Database
by Ali Al-Rubaie 16 Mar '09

16 Mar '09

Hi, Is it possible to access FS DB to retrieve data? Where can i find details about that? Thanks!

3 2