July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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?

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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

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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 >

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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

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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

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FreeSWITCH vs Yate
by Mike Fedyk 23 Jul '08

23 Jul '08

Hi, I've been using asterisk for a while now and just love running into assorted deadlocks, crashes, quirks and SIP misimplementations (ie, new Call-ID when forking/branching calls). I'm currently looking very closely at Freeswitch and another project that has been around longer, Yate. Yate seems to have a good architecture, and looks like it may be a good alternative to Freeswitch (if there was ever a need for it). Looking from the outside, it makes me wonder why Yate didn't just get a few more developers instead of another project being started instead. The Freeswitch devs may have looked at it before starting a new project and I'm curious as to why it was passed over. Could it be because Yate is written in C++? Personally since even now Freeswitch has a bigger community I'd probably only look at Yate to make sure I have a somewhat generic event model for Vicidial (somewhere on my to-do list). This message is by no means meant to be taken as an attack on Freeswitch. I'm just curious. Mike

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NAT traversal, SIP replies don't go to originating port
by Leon de Rooij 22 Jul '08

22 Jul '08

Hi all, I've set up Freeswitch, but my ATA can't register properly when it's behind NAT. Apparently, SIP replies aren't sent back to the port on the external IP of the NAT gw, where the initial request originated from. The situation is as follows: UA (172.31.0.55) ---- (172.31.0.1) NAT GW (1.1.210.76) ----- (1.1.232.18) FS The UA registers to a profile with these settings: <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="cust"/> <param name="dtmf-duration" value="100"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="1.1.232.18"/> <param name="sip-ip" value="1.1.232.18"/> <param name="sip-port" value="5060"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="disable-transcoding" value="true"/> <param name="manage-presence" value="true"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="disable-transfer" value="true"/> <param name="disable-register" value="false"/> <param name="tls" value="false"/> <param name="odbc-dsn" value="freeswitch:freeswitch:freeswitch"/> </settings> First I tried without STUN, and got the following trace: U 2008/07/22 21:03:48.983041 1.1.210.76:57501 -> 1.1.232.18:5060 REGISTER sip:test.nl:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.0.55:5060;branch=z9hG4bK57489228aa596f71. Max-Forwards: 70. To: <sip:ldr-line1@test.nl>. From: <sip:ldr-line1@test.nl>;tag=xETN4EDMxED. Call-ID: 928BD3011481111(a)172.31.0.55. CSeq: 2 REGISTER. Contact: <sip:ldr-line1@172.31.0.55:5060>. Content-Length: 0. Expires: 3600. . U 2008/07/22 21:03:48.983421 1.1.232.18:5060 -> 1.1.210.76:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 172.31.0.55:5060;branch=z9hG4bK57489228aa596f71;received=1.1.210.76. From: <sip:ldr-line1@test.nl>;tag=xETN4EDMxED. To: <sip:ldr-line1@test.nl>;tag=02F7cjtcp4cvS. Call-ID: 928BD3011481111(a)172.31.0.55. CSeq: 2 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9117M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="test.nl", nonce="d7ddacbe-5820-11dd- a225-93e83ac49152", algorithm=MD5, qop="auth". Content-Length: 0. . So, the reply sent from FS to the NAT gw, is sent to port 5060, while it originated from port 57501. Result is, that the '401 Unauthorized' never arrives at the ATA. Then I tried enabling STUN in the ATA, and got another result: U 2008/07/22 21:09:54.015734 1.1.210.76:61341 -> 1.1.232.18:5060 REGISTER sip:test.nl:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.210.76:59173;branch=z9hG4bK482f2934fda524ff. Max-Forwards: 70. To: <sip:ldr-line1@test.nl>. From: <sip:ldr-line1@test.nl>;tag=xIzM4EDMxID. Call-ID: 799E6DDE35D1111(a)1.1.210.76. CSeq: 1 REGISTER. Contact: <sip:ldr-line1@1.1.210.76:59173>. Content-Length: 0. Expires: 3600. . U 2008/07/22 21:09:54.015995 1.1.232.18:5060 -> 1.1.210.76:59173 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.1.210.76:59173;branch=z9hG4bK482f2934fda524ff. From: <sip:ldr-line1@test.nl>;tag=xIzM4EDMxID. To: <sip:ldr-line1@test.nl>;tag=2m2rg8UKgpS1g. Call-ID: 799E6DDE35D1111(a)1.1.210.76. CSeq: 1 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9117M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="test.nl", nonce="bfbe5042-5821-11dd- a225-93e83ac49152", algorithm=MD5, qop="auth". Content-Length: 0. . Now the reply is sent back to port 59173. That's the same as in the Contact as it's sent by the ATA. Does this mean STUN doesn't function properly ? I am using stun.fwdnet.net:3478 (ATA is a Zyxel P2002). Can I force FS to reply always to the port where the original message originated from ? Or should I fix this differently ? Thanks in advance, Leon de Rooij

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gen_mc: create Linksys certificates for SRTP
by Patrick 22 Jul '08

22 Jul '08

Hi, If you would like to test Linksys SRTP against FreeSWITCH then you'll probably bump into the problem of how to generate the required certificates as that is pretty much not documented. Apparently Linksys (Sipura) have an app called gen_mc (generate mini certificate) but they have not released that. I found a replacement on the Net, also called gen_mc. Haven't tried it yet and I don't know if the code can be trusted but maybe it's useful to others. The site where you can download the replacement gen_mc app: http://www.megajournal.ru/user/booBot/ Direct link to the file: http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c… (after download, remove the mp3, it's just a tar.gz) Regards, Patrick

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ANNOUNCEMENT: Weekly docs conference call, Wednesday UTC 1800/1pm Central
by Michael Collins 22 Jul '08

22 Jul '08

FYI, We've decided to go back to the originally scheduled time of 1pm Central. My apologies to those who cannot attend or who find this time inconvenient. We tried to find that one perfect time of day and day of week and it simply does not exist, so we're going with what works for the majority of those who handle the wiki documentation. The agenda for tomorrow is quite thin at the moment so please add some stuff: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_23 Also, even if you can't attend the meeting but there's stuff you want to see documented then please add your question/comment to the page. Thanks, Michael

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Question: PIKA cards in use?
by Michael Collins 22 Jul '08

22 Jul '08

FYI, Anthony needs to know if anyone has purchased a PIKA TDM card for use with FreeSWITCH. If you've bought one and are using it with FS then please let Anthony know about it. Thanks, MC

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