May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

calls appear to be dropping ... from landlines
by Dale Trub 02 Jun '09

02 Jun '09

We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729 We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring. Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us? Is there some SIP logging we can do to debug? Are there any paid contractors avail who would have the expertise to look into this? Any help appreciated ... this is a major issue for us! Thanks much, -Dale daletrub(a)gmail.com

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Can Freeswitch + LAMP run on 128MB RAM?
by Fred-145 02 Jun '09

02 Jun '09

Hello Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB NAND flash. Before I go ahead and check, would someone know if a minimal Linux + LAMP server* + Freeswitch can run OK with this amount of memory? Thank you. * I think I'll trade MySQL with Firebird, to avoid buying a license for my commercial application -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Conference users hear MOH until leader enters?
by Jason Cromes 02 Jun '09

02 Jun '09

First off, I apologize if this has been sent multiple times, the mailing list won't cooperate with me... Hopefully that is resolved now. I'm attempting to replicate the behavior of an Asterisk conferencing system and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily) in FreeSwitch. I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the prompts and conferencing setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database table, but I need them disabled until the conference owner dials in and enables his conference. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. (The purpose of this is to prevent abuse of conferences that are, basically, always available.) This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, that user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks...

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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?
by Peter Olsson 01 Jun '09

01 Jun '09

I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Michael Jerris [mike(a)jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I’m on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Nope – it’s not :) Just to make sure I even deleted the source completely, and checked everything out again. Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682!

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Rotating log files not working
by Lars Zeb 01 Jun '09

01 Jun '09

I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command 'kill -s HUP pid' (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings> <param name="default-template" value="sql"/> <param name="rotate-on-hup" value="true"/> <param name="legs" value="a"/> </settings> . What am I doing wrong here? Thanks, Lars

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g729 support
by FERNANDO VILLARROEL 01 Jun '09

01 Jun '09

Dear All, In this moment i am using Asterisk for my services VoIP; i am testing FS and i am very interest on change Asterisk for FS, my motivation is improve my qualify mainly ACD statics. But i have problem, G729 codec is only supported in passthrough mode. FreeSwitch does not do any transcoding with G729 yet, hence I can't fully move to FS without G729 support. I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to receive this traffic and convert to ulaw and forward to other SIP provider, how i can do or anyone help me? I hope your comments or idea how i can do, Fernando.

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Help with lua and channel variable processing
by Larry Marshall 01 Jun '09

01 Jun '09

I would like to process the ${billsec} variable in the conf/autoload_configs/cdr_csv.conf.xml file below, returning a different value for the INSERT command. I already have written a lua function which takes an integer value from ${billsec} and returns a string. It is stored in freeswitch/scripts as sec2time.lua. Do I change the sql template below to ., ${lua(billsec)}, .? <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings> <param name="default-template" value="sql"/> <param name="rotate-on-hup" value="true"/> <param name="legs" value="a"/> </settings> <templates> <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${contex t}" ,"${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec} ",... </templates> </configuration> Thanks, Lars

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JavaScript apiExecute vs session.execute
by jcromes＠gmail.com 31 May '09

31 May '09

From JavaScript, what is the difference between apiExecute and session.execute? It seems they are mostly interchangable, in fact the Wiki gives examples of them doing identical operations. What advantages does one have over the other? Are there some things that cannot or should not be called from one or the other? Thanks!

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Is still Woomera supported again?
by Sadjad Seyed-Ahmadian 31 May '09

31 May '09

Hi dear All, I tried to support H323 by opal thaat failed. I want to know is still Woomera supported by FreeSwitch or not. Sincerely, Sadjad

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Missing Events in mod_event_socket
by Gerry Hull 31 May '09

31 May '09

Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: "It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it." Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine. However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry

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