June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can freeswitch work with ODBC for Microsoft Access files?
by Frank Church 05 Jun '10

05 Jun '10

I'd just like to know if any brave souls have tested Freeswitch with Microsoft ODBC for JET 4.0 using Microsoft Access or any of the other file formats it supports, like DBF and Paradox db files. Anyone? -- Frank Church ======================= http://devblog.brahmancreations.com

1 1

Question regarding table channels in core.db
by DJB INTERNATIONAL 05 Jun '10

05 Jun '10

I would like to know whether this is the normal behavior when i ran a query during the call for table channels: select * from channels I noticed that the ip_addr for inbound and outbound shows the same value. Should it show the bleg_ip_addr for outbound direction instead? git-0152706 2010-06-04 19-37-04 -0500 Thank you, Dorn B.

1 0

Re: [Freeswitch-users] Set Privacy:id on Outbound Call
by Michael De Lorenzo 05 Jun '10

05 Jun '10

I tried setting the sip_h_Privacy via: freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. cid .. ",sip_h_Privacy=id,ignore_early_media=true}sofia/gateway/mygateway/" .. call_this_number) but in the SIP Capture Privacy is still none: INVITE sip:9735551234@5.6.7.8 <sip%3A9735551234(a)5.6.7.8> SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5K Max-Forwards: 70 From: "Anonymous" <sip:9727289377@1.2.3.4 <sip%3A9727289377(a)1.2.3.4> ;transport=udp>;tag=QjjNSt2m3g0rB To: <sip:9735551234@5.6.7.8 <sip%3A9735551234(a)5.6.7.8>> Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9 CSeq: 131718385 INVITE Contact: <sip:gw+verizon1@1.2.3.4:5080;transport=udp;gw=verizon1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "Anonymous" <sip:+19727289377@1.2.3.4<sip%3A%2B19727289377(a)1.2.3.4> > v=0 o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 27884 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Am I doing something incorrectly?

2 1

Presence/BLF
by Troy Anderson 05 Jun '10

05 Jun '10

I am trying to get my head around presence/BLF indicators in a mixed endpoint environment. We have a couple Cisco 7940's and a couple Aastra phones (55i and 57i), and some Polycoms (450 and others). Presence works great if I am just using the Aastras, but gets wonky when I mix in other phones. I am using Wireshark to try to understand what's going on, but need some advise on what to look for. Is all this handled with SIP NOTIFY events? Is FS completely in charge of the BLF status on the phones? Thanks for any pointers! -Troy

3 2

Help debugging INVITE with provider
by François Legal 04 Jun '10

04 Jun '10

Hello, I'm having trouble to send/receive calls with one of my providers. Freeswitch says the reply sent by provider has the session information incorrect, then the provider replies the calle is busy (which of course is not). The sip trace follows. Anybody can help ? François send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------

4 7

check status of a number
by Madovsky 04 Jun '10

04 Jun '10

Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck

3 10

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

1 0

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

1 0

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