- FreeSWITCH-users - FreeSWITCH Mailing Lists

ring groups
by Star Wind 24 Jul '08

24 Jul '08

Who can tell me, how I can create a static ring group? I'm using third-party SBC, which changes "Contact" field on every registration, every 15 minutes. That's why I can't use "add-group" and "del-group" dialplan features, because new "Contact" field generates randomly. All I need - simply ring group, with ring scheme like this: <phone 1 rings...> - timeout - <phone 2 rings...> - timeout - ... and so on.

4 6

Timeout parameter in originate
by Ashutosh 24 Jul '08

24 Jul '08

Hi, We see originate function as follows: result = new_session.originate(session, dest[[[[[[[[[[, dialplan], context], cid_name], cid_num], network_addr], ani], aniii], rdnis], username], to]); Where to= timeout When does the timeout counter start, after call to the originate Api, or after the remote end has picked up the call ? Thanks, ashutosh

1 0

Operator/Receptionist console
by Brian Snipes 24 Jul '08

24 Jul '08

Ok.. I need some suggestions. Currently I have a couple of snom 370 phones with 2 sidecars on each phone for the receptionist and her backup location. Those phones monitor the status of ~55 extensions. The Snoms seem to be having a devil of a time doing this. Several times a day they get REALLY slow. This happens consistently also right after the phone is rebooted and the subscriptions start showing up. By slow, I mean I can hit 3 or 4 keys before anything happens on the display and when it starts filling in the display it is slow. If I remove all subscription monitoring from the phones they speed along like I would expect them to do. Pulling on all of your vast wisdom - what options do I have? What phones will work properly with that many subscriptions in a sip environment? On another note - I am thinking about creating an application similar to FOP for the desktops and need suggestions on the best way to do this. This won't be anything fancy. I only need it to alert whether a phone is in use or not. Here are the ideas so far: 1. Query existing call db (maybe the limit table) every minute. Dump results of extensions in use to an html file. Clients view html file and it is set to refresh every 15 seconds. < I don't really like this too much > 2. Have something similar to another product I've used. What it did was send events to a modified irc daemon and a windows client listened to the channel and updated the extension buttons with their current status. Thanks! Brian

4 8

safe_freeswitch (like safe_asterisk): restarting FS automatically?
by Birgit Arkesteijn 24 Jul '08

24 Jul '08

Hi all, We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux x86_64 machine. At the moment it's crashing few times a day, making our services very unreliable. At the moment we don't have the time to rebuild this version, so I'm looking for an equivalent of the "safe_asterisk" script. This script runs Asterisk in a loop, restarting it when it goes down. I couldn't find any equivalent script, but maybe I using the wrong keywords in my search. Does anyone know if such a script is available, and (if so) could you point out where I can find it? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

6 9

Documentation questions
by Michael Collins 23 Jul '08

23 Jul '08

Guys, I was just looking over the source directory an I noticed that the common files for an open source project may need some serious attention: AUTHORS INSTALL NEWS README I'm sure that the COPYING file is just fine. NEWS and README are completely empty. Should we maybe just put references in them that point to interesting spots on the freeswitch.org site? Does anyone have suggestions for content in these files? Should we delete them altogether? INSTALL basically just says to do the "./config && make && make install" shuffle. Is that adequate or should we possibly add a link to the GSG on the wiki? AUTHORS looks pretty good except that maybe BKW should be moved up under MikeJ and referenced as the tester/QA/etc. person for FS. Also, for those who have contributed in the past but who are not current contributors should we have a separate section that says something like, "We'd like to thank the past contributors for their efforts:" and then list guys like Bret McDanel and Josh Culp? Last question - could those in the know look to see if there are any glaringly obvious omissions? As far as OpenZAP goes I will work on these same files and come up with the requisite additions. (I think I'm unofficially the official OZ documentation guy.) Please give us your $0.02 so we can act accordingly. Thanks, MC

1 0

sample rate with record_session
by Frank - IMPACT 23 Jul '08

23 Jul '08

Running FreeSwitch Version 1.0.trunk (9111) on RH 8 Record_session works fine with this <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_n umber}.wav"/> But the sample rate and resulting wav file are way more information than I need. I tried to half the sample rate with this inserted before we run record_session. <action application="set" data="record_rate=5512"/> But that did not seem to reduce the sample rate and the resulting wav file. Is this the wrong variable to set or is this not possible to reduce sample rates and wav size files?

3 3

Equivalent of asterisk ID
by Ashutosh 23 Jul '08

23 Jul '08

Hi, I have been looking for an attribute in the FS CDR which will be unique for a call, like we had uniqueid field in asterisk cdr. Someone have any idea ? Thanks, ashutosh

5 9

Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration
by Diego Viola 23 Jul '08

23 Jul '08

Hi, I'm sorry for my reaction... I know FreeSWITCH is better compared to Asterisk, and I will learn it, with XML or not. I also don't think XML is a bad thing, it's an open standard and I think I will get used to it. @Ognjen, when you say: "You can't be serious that Asterisk/Digium is ok when you ask them for a feature or anything at all? Tony offered them to rewrite obvious broken core to make it better, they said no. I tried 1.4 because of the new features advertised, and I got no sleep beacuse of it for two weeks. What's the point of polishing something that is without any doubt broken in core? I don't understand why are you even looking at FS as an option, if Asterisk is ok for your needs?" What do you mean with "broken core" exactly? I know asterisk has problems but I would like to know what core problems you refer there. I'm just curious about it. I'm looking at FS as an alternative to Asterisk because I been having a lot of problems with Asterisk, and I want a system that works better, the problems I had are the following: http://bugs.digium.com/view.php?id=12628 http://bugs.digium.com/view.php?id=12653 http://bugs.digium.com/view.php?id=12810 And I had more than that... and I keep having problems, I want to move on and have a system that I can trust. Anyway... I will learn FS and XML, and sorry for my reaction... I know Brian is a good and cool guy and he is the guy that introduced me to FS. Peace out, Diego On Sun, Jun 29, 2008 at 10:20 PM, Michael Collins <mcollins(a)fcnetwork.com> wrote: >> This XML argument has gone on since FreeSwitch was unveiled and quite >> frankly I was one of the people railing on Tony for that decision, but > as >> I >> have continued to look at the core and develop soluitons on this > platform >> it >> has become more and more obvious that I was wrong initially... (wow >> someone >> archive tha.. Its the only time you'll see it on a mailing list :P ) >> K > > > So noted! It would be nice to get an audio recording of you groveling > for Tony's forgiveness... we could add it to the demo IVR: > > "...To hear Ken Rice admit the folly of his ways, please press ..." > > :) > > -MC > > P.S. - Can you record that in UWB mode? =D >

9 12

FreeSWITCH in the press...
by Brian West 23 Jul '08

23 Jul '08

Please check this out and digg the article http://digg.com/software/Open_Source_VoIP_Asterisk_or_FreeSwitch Brian West sip:brian@freeswitch.org

3 2

Freeswitch Performance Testing
by Juan Backson 23 Jul '08

23 Jul '08

Hi, We are running some stress tests again freeswitch using sipp. At 100 concurrent call, Freeswitch starts to throw out some strange error messages: freeswitch@lab> 2008-07-21 06:27:51 [CRIT] switch_core_session.c:826 switch_core_session$ 2008-07-21 06:27:51 [CRIT] sofia.c:3177 sofia_handle_sip_i_invite() LUKE: I'm hit, but not $ 2008-07-21 06:27:51 [CRIT] sofia.c:3178 sofia_handle_sip_i_invite() LUKE'S VOICE: Artoo, se$ Green laserfire moves past the beeping little robot as his head turns. After a few beeps a$ Artoo reduces the max sessions to 328 thus, saving the switch from certain doom. ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2008-07-21 06:41:00:154 1216593660.154593 Last Reset Time | 2008-07-21 06:49:25:136 1216594165.136188 Current Time | 2008-07-21 06:49:26:139 1216594166.139989 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:01:003 | 00:08:25:985 Call Rate | 9.970 cps | 9.998 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 10 | 5059 Total Call created | | 5059 Current Call | 100 | -------------------------+---------------------------+-------------------------- Successful call | 10 | 4387 Failed call | 0 | 572 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:018 | 00:00:08:545 Call Length | 00:00:10:021 | 00:00:22:125 ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic ----- Last Error: Aborting call on unexpected message for Call-Id '3143-6125@1... When increasing the number of concurrent calls to several hundreds, Freeswitch throws the following error: [CRIT] switch_time.c:394 softtimer_runtime() Over Session Rate of 30! The test is run on Dual Core Intel P4 3GHz with 1G of RAM. Is this kind of performance normal for such kind of machine or there are some OS configs required to enable more concurrent calls? What kind of number should I be expecting to get? JB

6 10