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BerkeleyTIP Join June Global Free SW HW Culture Mtgs via VOIP or in Berkeley
by giovanni_re 04 Jun '10

04 Jun '10

You're invited to join in with the friendly people at the BerkeleyTIP global meeting - newbie to Ph.D. - everyone is invited. Get a headset & join using VOIP online, or come to Berkeley. 1st step: Join the mailing list: http://groups.google.com/group/BerkTIPGlobal Watch the videos. Discuss them on VOIP. 8 great videos/talks this month - see below. Starting off year 3 of BerkeleyTIP :) Join with us at the LOCATION TO BE DETERMINED at the University of California at Berkeley, or join from your home via VOIP, or send this email locally, create a local meeting, & join via VOIP: Tip: a wifi cafe is a great place to meet. :) JUNE 5 & 20 AT UCB MEETING LOCATIONS TO BE DETERMINED. PLEASE VIEW THE BTIP WEBSITE & MAILING LIST PAGES FOR THE LATEST DETIALS. http://sites.google.com/site/berkeleytip http://groups.google.com/group/BerkTIPGlobal BerkeleyTIP - Educational, Productive, Social For Learning about, Sharing, & Producing, All Free SW HW & Culture. TIP == Talks, Installfest, Project & Programming Party http://sites.google.com/site/berkeleytip ===== CONTENTS: 1) 2010 JUNE VIDEOS; 2) 2010 JUNE MEETING DAYS, TIMES, LOCATIONS; 3) LOCAL MEETING AT U. C. Berkeley LOCATION TO BE DETERMINED; 4) HOT TOPICS; 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) ; 6) INSTALLFEST; SPECIAL: FREE 7th Annual BERKELEY WORLD MUSIC FESTIVAL June 5 Sat; 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN; 8) IRC: #berkeleytip on irc.freenode.net; 9) VOIP FOR GLOBAL MEETING; 10) VOLUNTEERING, TO DOs; 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce; 12) ANYTHING I FORGOT TO MENTION?; 13) FOR FORWARDING ======================================================================= ===== 1) 2010 JUNE VIDEOS Super Computing for Business, Brian Modra, CLUG State of the Linux Union 2010, Jim Zemlin, Linux Foundation Journaled Soft-Updates, Dr. Kirk McKusick, BSDCan 2010 Scientific data visualization using Mayavi2, Gael Varoquaux, Python4ScienceUCB Bringing OLPC to children in Afghanistan, Carol Ruth Silver, OLPC-SF Text-to-Speech in Ubuntu with Kttsd Kmouth Festival, blip.tv Rabbi Rabbs, the UnixRabbi, leads a group of Unix geeks, Comedy, UUASC, BS"D, 2003 The Great Debate - Are We Alone?, Geoff Marcy and Dan Werthimer, SETI@UC Berkeley Thanks to all the speakers, organizations, & videographers. :) [Please alert the speakers that their talks are scheduled for June for BTIP (if you are with the group that recorded their talk), because I may not have time to do that. Thanks. :) ] URLs for video download & full details: http://sites.google.com/site/berkeleytip/talk-videos Download & watch these talks before the BTIP meetings. Discuss at the meeting. Email the mailing list, tell us what videos you'll watch & want to discuss: http://groups.google.com/group/BerkTIPGlobal Know any other video sources? - please email me. _Your_ group should video record & post online your meeting's talks! ===== 2) 2010 JUNE MEETING DAYS, TIMES, LOCATIONS In person meetings on 1st Saturday & 3rd Sunday, every month. June 5 & 20, 12N-3P USA-Pacific time, Saturday, Sunday Online only meeting using VOIP: June 14 & 29, 5-6P USA-Pacific time, Monday, Tuesday Mark your calendars. 5 Sat 12N-3P PDST = 3-6P EDST = 19-22 UTC 14 Mon 5-6P PDST = 8-9P EDST = 0- 1 UTC Tues 15 20 Sun 12N-3P PDST = 3-6P EDST = 19-22 UTC 29 Tues 5-6P PDST = 8-9P EDST = 0- 1 UTC Wed 30 USA-PacificDaylightSavingsTime is -7 hours UTC, due to daylight savings currently. Times listed above should be double checked by you for accuracy. ===== 3) LOCAL MEETING AT U. C. BERKELEY - LOCATION TO BE DETERMINED http://sites.google.com/site/berkeleytip/directions RSVP please. See below. It greatly helps my planning. But, _do_ come if you forgot to RSVP. THE JUNE 5 & 20 ON CAMPUS MEETING LOCATIONS ARE CURRENTLY TO BE DETERMINED. HOPEFULLY KNOW ON FRIDAY JUNE 4. PLEASE VIEW THE BTIP WEBSITE & MAILING LIST FOR THE FINALIZED LOCATION. THANK YOU. ALWAYS BE SURE TO CHECK THE BTIP WEBSITE _&_ MAILING LIST FOR THE LATEST LAST MINUTE DETAILS & CHANGES, BEFORE COMING TO THE MEETING! :) http://sites.google.com/site/berkeleytip http://groups.google.com/group/BerkTIPGlobal DO BRING A VOIP HEADSET, available for $10-30 at most electronics retail stores, & a laptop computer, so you are able to communicate with the global BTIP community via VOIP. It is highly recommended that you have a voip headset, & not rely on a laptop's built in microphone & speakers, because the headphones keep the noise level down. Bringing a headset is not required, but is a great part of the being able to communicate with the global community. :) Clothing: Typically 55-80 degrees F. Weather: http://www.wunderground.com/auto/sfgate/CA/Berkeley.html Other location local meeting possibilities: http://sites.google.com/site/berkeleytip/local-meetings Create a local meeting in your town. ===== 4) HOT TOPICS Oracle owns Sun - Free SW implications? OpenOffice? OpenOffice - Ready for college Fall 2010? MakerBot, RepRap - Personal Making - Like where PCs were in 1975? Android phones - Besting iPhone? worthwhile? How knowable is the hw? - Can BSD be run on Android phones? iPad, iPhone & iPod- rooting & running GNU(Linux) & BSD ===== 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) If you think there is a >70% chance ("likely") you'll come to the in person meeting in Berkeley, please RSVP to me. Thanks. It helps my planning. Please _do_ come even if you haven't RSVP'd, it's not required. Better yet, join the BerkeleyTIP-Global mailing list, send the RSVP there, & tell us what things you're interested in, or what videos you'll try to watch - so we can know what videos are popular, & we might watch them too. :) http://groups.google.com/group/BerkTIPGlobal ===== 6) INSTALLFEST Get help installing & using Free Software, Hardware & Culture. Laptops only, typically. There isn't easy access for physically bringing desktop boxes here. RSVP _HIGHLY RECOMMENDED_ if you want installfest help. Please RSVP to me, Giovanni, at the from address for this announcement, or better, join & send email to the BTIP-Global mailing list telling us what you'd like help with. This way we can be better prepared to help you, & you might get valuable advice from the mailing list members. If you are new to using free software, an excellent system would be the KUbuntu GNU(Linux) software. It is very comprehensive, fairly easy to use (similar to Windows or Mac), & suitable for personal, home, university, or business use. We are also glad to try to help people with software who join via VOIP. Please email the mailing list with requests that you want help with, so we can try to be prepared better to help you. Installfest volunteers/helpers always welcome, in person, or via VOIP. :) ===== SPECIAL - MUSIC IN BERKELEY JUNE 5 SATURDAY ===== Also June 5 Sat in Berkeley, after BTIP: ===== FREE 7th Annual BERKELEY WORLD MUSIC FESTIVAL ===== Telegraph Avenue 12 Noon - 9 pm "Berkeley World Music Fest... has some of our best world musicians who call East Bay home...a wonderful event, and intimate, mix of outdoor performances in cafes and shops(the best music People's Park gets each year along Telegraph Ave. (Larry Kelp, KPFA music host)" Continuous music outdoors & in cafes http://www.berkeleyworldmusic.com/entry.asp?PageID=118 ===== 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN If you get to the meeting & don't see a "BerkeleyTIP" sign up yet, please: 1) Make a BTIP sign on an 8x11 paper & put it at your table, 2) Email the mailing list, or join on IRC, & let us know you are there. Ask someone if you could use their computer for a minute to look something up, or send an email. People are usually very friendly & willing to help. We can also email you a temporary guest AirBears account login. We will have wifi guest accounts available for BTIP attendees. Be sure you have wifi capable equipment. Be Prepared: Bring a multi-outlet extension power cord. ===== 8) IRC: #berkeleytip on irc.freenode.net For help with anything, especially how to get VOIP working, & text communication. ===== 9) VOIP FOR GLOBAL MEETING Speak & listen to everyone globally using VOIP. Get a headset! See some VOIP instructions here: http://sites.google.com/site/berkeleytip/voice-voip-conferencing ===== 10) VOLUNTEERING, TO DOs Enjoy doing or learning something(s)? Help out BTIP in that area. Website development, mailing list management, video locating, VOIP server (FreeSwitch, Asterisk) or client (Ekiga, SFLPhone,...), creating a local meeting. Join the mailing list & let us know. Your offers of free help are always welcome here. :) ===== 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce Everyone should join the BerkeleyTIP-Global list: http://groups.google.com/group/BerkTIPGlobal Say "hi", tell us your interests, & what videos you'll like to watch. Info on all lists here: http://sites.google.com/site/berkeleytip/mailing-lists ===== 12) ANYTHING I FORGOT TO MENTION? Please join & email the BerkeleyTIP-Global mailing list. ===== 13) FOR FORWARDING You are invited to forward this message anywhere it would be appreciated. Better yet, use it to create a local meeting. Invite & get together with your friends locally, & join in with us all globally. :) Looking forward to meeting with you in person, or online. :) Giovanni

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Unable to detect the digits in demo IVR when connecting via DID inbound
by Russell Kwok 04 Jun '10

04 Jun '10

Hello all, Sorry to bother all, I had been trying so hard to create a DID-inbound that will route to demo IVR(5000 XML default) and allow user to press and select the menu via pressing digits. It works perfect when I dial 5000 from any of my local extension. However, when I try to call the DID, and got connected to the voice mail successfully, but the digit is unable to be detected. Wondering if my configuration got something wrong? what i had created in the public\01_mydivert.xml (i put the config inside the public context and setting mydivert to connect to my pbxserver at port 5080) file is as follow: <extension name="didfromMyDivert"> <condition field="network_addr" expression="^78\.46\.23\.202$" > <action application="info" /> <action application="set" data="domain_name=$${domain}" /> <action application="set" > data="transfer_ringback=$${uk_ring}" /> <action application="transfer" data="5000 XML default" /> </condition> </extension> and when i call the DID number hosted by mydivert.com, the log file shows when i press 1, it just do nothing and keep playing to "please press 2 or something" 2010-06-04 12:23:59.414516 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:23:59.414516 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:23:59.814618 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:23:59.933629 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/1.wav] (en:en) 2010-06-04 12:23:59.933629 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:24:00.394005 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:00.514102 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-to_do_a_freeswitch_echo_test.wav] (en:en) 2010-06-04 12:24:00.514102 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms > 2010-06-04 12:24:01.596587 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:1280 2010-06-04 12:24:01.599459 [DEBUG] zap_io.c:2061 4:1 GENERATE DTMF [1] 2010-06-04 12:24:01.639329 [DEBUG] mod_openzap.c:721 queue DTMF [1] > 2010-06-04 12:24:01.654270 [DEBUG] switch_rtp.c:1928 Send start packet for > [1] ts=172800 dur=160/160/2000 seq=63824 2010-06-04 12:24:01.674124 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=320/320/2000 seq=63825 2010-06-04 12:24:01.694115 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=480/480/2000 seq=63826 2010-06-04 12:24:01.714098 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=640/640/2000 seq=63827 2010-06-04 12:24:01.733685 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=800/800/2000 seq=63828 2010-06-04 12:24:01.754129 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=960/960/2000 seq=63829 2010-06-04 12:24:01.774264 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1120/1120/2000 seq=63830 2010-06-04 12:24:01.795396 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1280/1280/2000 seq=63831 2010-06-04 12:24:01.814112 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1440/1440/2000 seq=63832 2010-06-04 12:24:01.834315 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1600/1600/2000 seq=63833 2010-06-04 12:24:01.853792 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1760/1760/2000 seq=63834 2010-06-04 12:24:01.874010 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1920/1920/2000 seq=63835 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63836 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63837 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63838 2010-06-04 12:24:02.333695 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:02.434126 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-please.wav] (en:en) 2010-06-04 12:24:02.434126 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:24:02.854127 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:02.974217 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:24:02.974217 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:24:03.374396 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:03.494367 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/2.wav] (en:en) 2010-06-04 12:24:03.494367 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms and the app info for this DID inbound is as follow: 2010-06-04 12:23:41.001560 [INFO] mod_dptools.c:973 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/24172200(a)78.46.23.202] Unique-ID: [f456ff9c-6f90-11df-84c8-3b7d920feee8] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [24172200] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [24172200] Caller-Caller-ID-Number: [24172200] Caller-Network-Addr: [78.46.23.202] Caller-ANI: [24172200] Caller-Destination-Number: [85236931541] Caller-Unique-ID: [f456ff9c-6f90-11df-84c8-3b7d920feee8] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/24172200(a)78.46.23.202] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1275625420968944] Caller-Channel-Created-Time: [1275625420968944] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [false] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_direction: [inbound] variable_uuid: [f456ff9c-6f90-11df-84c8-3b7d920feee8] variable_sip_local_network_addr: [192.168.8.232] variable_sip_network_ip: [78.46.23.202] variable_sip_network_port: [5060] variable_sip_received_ip: [78.46.23.202] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [24172200] variable_sip_from_uri: [24172200(a)78.46.23.202] variable_sip_from_host: [78.46.23.202] variable_sip_from_user_stripped: [24172200] variable_sip_from_tag: [as750696c8] variable_sofia_profile_name: [external] variable_sip_Remote-Party-ID: ["24172200" <sip:24172200@78.46.23.202<sip%3A24172200(a)78.46.23.202> > >;privacy=off;screen=no] variable_sip_cid_type: [rpid] variable_sip_full_via: [SIP/2.0/UDP 78.46.23.202:5060 > ;branch=z9hG4bK32bcbf22] variable_sip_from_display: [24172200] variable_sip_full_from: ["24172200" <sip:24172200@78.46.23.202<sip%3A24172200(a)78.46.23.202> > >;tag=as750696c8] variable_sip_full_to: [<sip:85236931541@jessnruss.dyndns.org:5080>] variable_sip_req_user: [85236931541] variable_sip_req_port: [5080] variable_sip_req_uri: [85236931541@jessnruss.dyndns.org:5080] variable_sip_req_host: [jessnruss.dyndns.org] variable_sip_to_user: [85236931541] variable_sip_to_port: [5080] variable_sip_to_uri: [85236931541@jessnruss.dyndns.org:5080] variable_sip_to_host: [jessnruss.dyndns.org] variable_sip_contact_user: [24172200] variable_sip_contact_uri: [24172200(a)78.46.23.202] variable_sip_contact_host: [78.46.23.202] variable_channel_name: [sofia/external/24172200(a)78.46.23.202] variable_sip_call_id: [6b53813d4bb32def4d2931960d17e7ee(a)78.46.23.202] variable_sip_user_agent: [Asterisk] variable_sip_via_host: [78.46.23.202] variable_sip_via_port: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=root 1913419625 1913419625 IN IP4 78.46.23.202 s=Asterisk PBX 1.6.0.26 c=IN IP4 78.46.23.202 t=0 0 m=audio 19066 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ] variable_remote_media_ip: [78.46.23.202] variable_remote_media_port: [19066] variable_sip_use_codec_name: [PCMU] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] However, if i call the demo IVR 5000 internally, the log shows really good. 2010-06-04 12:30:22.394100 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:30:22.394100 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:30:22.794457 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:22.914491 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/1.wav] (en:en) 2010-06-04 12:30:22.914491 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms 2010-06-04 12:30:23.373688 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:23.474363 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-to_do_a_freeswitch_echo_test.wav] (en:en) 2010-06-04 12:30:23.474363 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16@8000hz 1 channels 20ms > 2010-06-04 12:30:25.215382 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:960 2010-06-04 12:30:25.215382 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:25.334284 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 > digits t/o 2000 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:376 digits '1' 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:470 action regex [1] > [/^(10[01][0-9])$/] [0] > 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:488 IVR action on menu > 'demo_ivr' matched '1' param 'bridge sofia/ > 192.168.8.232/888(a)conference.freeswitch.org' > 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:492 > switch_ivr_menu_execute todo=[2] EXECUTE sofia/internal/1001(a)jessnruss.dyndns.org bridge(sofia/ > 192.168.8.232/888(a)conference.freeswitch.org) from the two logs above, what i can see the difference is the line: in [FAILURE log] 2010-06-04 12:24:01.596587 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:1280 in [SUCCESS log] 2010-06-04 12:30:25.215382 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:960 i am wondering if is there anything to do with this line? any help is really appreciated. Cheers, Russell Kwok

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Question regarding codec param
by DJB INTERNATIONAL 03 Jun '10

03 Jun '10

I would like to know whether when I set disable-transcoding=true, would it be the same as setting inherit_codec = true? Thank you.

2 1

Spikes in Load on Ubuntu
by Matthew Fong 03 Jun '10

03 Jun '10

I'm running 1.0.6 and I noticed that right after upgrading from 15135 that the load (15 min avg) on my system seems to have increased about double running the same amount of traffic. I've been running top all day for the past few days and I noticed that for the most part I always have 75% idle cpu, but maybe once ever 45 minutes or an hour there is an spike the 1 minute load averages (maybe up to 9). This is only a dual core processor so I believe 9 usually indicates that are some processes that are waiting. The idle % does not change, and I have free memory, but for whatever reason the linux load algorithm spikes (maybe very high for like 1-5 seconds). At first I thought this might be related to a tickless timer, so I disable it from the kernel, but today I am still getting the same spikes. So my question is, are these spikes something to worry about, or should I just monitor the % of idle cpu available on my box for guesstimating how much further I can push this system? Thanks. --matt

2 1

Help get freeswitch.com back
by Anthony Minessale 03 Jun '10

03 Jun '10

Some squatter has had freeswitch.com for years and it's time we had it back. We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute. They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action) Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause. Everyone who donates will get their name up on our thank you page on our site. Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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OSTAG - Open Source Telephony Action Group
by Michael Collins 03 Jun '10

03 Jun '10

Good news! The Open Source Telephony Action Group (OSTAG) is almost ready to become an official 501(c)3 non-profit organization. We need to raise $800 in order to file the necessary paperwork. We have set up a PayPal account where we can all send our donations: donations(a)ostag.org Please assist us in moving this important organization forward. OSTAG is dedicated to the advancement of open source telecommuncations software that improves the lives of people all over the world. This is a most worthy cause! Please help now and in the future. -Michael

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Out-Going Call Transfer Question
by Michael Gende 03 Jun '10

03 Jun '10

Hello, I wonder if someone could direct me to some documentation (I'm sure it exists) on how to undertake something. We've been happily using FS for some time now. Our is set-up emulates a pretty standard key system. To the point of my post: I can, for incoming calls, easily and successfully transfer them to conference rooms and other registered extensions. However, if I call out to another party, I can only transfer the call to a conference room, not another extension. I suspect this has to do with FS not knowing - or me not correctly telling it how - to handle the "out going" case. I.e, which "end" of the call to transfer. Sometimes its handy to make a call and transfer the callee to some other extension. Any suggested reading? Thanks, Mike G.

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xml_cdr http post differentiation between legs
by Adam Ford 03 Jun '10

03 Jun '10

Is there any way to tell whether it is the a-leg or b-leg when receiving the POST data from mod_xml_cdr? I wrote a parser in Python which parses the a_*.cdr.xml log files, and only grabs the info in the associated b-leg log files if it needs additional information(for example - in the case of a call that ends with mod_fifo). I want to adapt this to using the POST features of mod_xml_cdr so that it can be real time. Any suggestions are greatly appreciated, -Adam

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Status of absolute_codec_string
by Kristian Kielhofner 03 Jun '10

03 Jun '10

Hello everyone, Either I am misunderstanding absolute_codec_string or there is something going on with it in trunk. I had a git checkout from a month ago that exhibited this same behavior so I updated about an hour ago. It's still there (or I'm still confused). In short, I have a call that comes in from a remote endpoint that will advertise PCMU and G729 (in that order). When I bridge the call I want to remove G729 capability from the outbound leg in some cases so FS doesn't have to transcode. In other cases (controlled from the dialplan), I want to place G729 as the first codec (again so FS doesn't have to transcode) but I can force G729. I've tried regex on ep_codec_string and I've tried regex on the SDP. Both seemed to work well but what didn't work was setting absolute_codec_string from either of these. Even when I try to set it manually without any fancy conditions it doesn't appear to work. The codec prefs from the SIP profile are presented to the remote endpoint. Full debug here: http://pastebin.freeswitch.org/13088 Any ideas? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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xml_curl endpoint register
by Gustavo Espeche 03 Jun '10

03 Jun '10

Hello all, i'm new in freeswitch we are migrating our software form yate to freeswitch. I'm writing a registration php script using xml_curl module, the registration was good but some endpoint don't use 5060 as sip port and when i register it, i can't find the endpoint port in curl post, because of then when i dial to an endpoint that isn't listen in sip port 5060 the call fail. Some one know how can i fix it? when i register the endpoint i insert in a mysql db time, ip, and when some one dial to this user i read my db for know the user's ip. but i need insert the endpoint sip's port too for it work. Best Regards. Gustavo Espeche www.easyipcall.com

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