August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] upadte FS (make current)
by Peter Olsson 07 Aug '12

07 Aug '12

Usually it's no problem. However, I always recommend to have a test server to run basic tests before going into production. There have been a couple of memory leaks fixed, but I think that was before your current version - but I don't remember all the commits from my head.. :) I would also suggest to read through the git commit log, to get an idea of what has been done. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Miha Skickat: den 7 augusti 2012 10:37 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] upadte FS (make current) Hi, I am running on production server FS (runs ok). I would like to upgrade it to latest git. Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr). Is it better to first test everything on test server and than make upgrade? My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? : top - 09:46:46 up 13 days, 12:32, 3 users, load average: 0.00, 0.04, 0.05 Tasks: 159 total, 1 running, 158 sleeping, 0 stopped, 0 zombie Cpu(s): 0.6%us, 0.4%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4046636k total, 3356340k used, 690296k free, 357296k buffers Swap: 6094840k total, 0k used, 6094840k free, 2629280k cached My version: FreeSWITCH Version 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5 (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z) Thanks! Miha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5020d1a332763868583022!

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upadte FS (make current)
by Miha 07 Aug '12

07 Aug '12

Hi, I am running on production server FS (runs ok). I would like to upgrade it to latest git. Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr). Is it better to first test everything on test server and than make upgrade? My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? : top - 09:46:46 up 13 days, 12:32, 3 users, load average: 0.00, 0.04, 0.05 Tasks: 159 total, 1 running, 158 sleeping, 0 stopped, 0 zombie Cpu(s): 0.6%us, 0.4%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4046636k total, 3356340k used, 690296k free, 357296k buffers Swap: 6094840k total, 0k used, 6094840k free, 2629280k cached My version: FreeSWITCH Version 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5 (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z) Thanks! Miha

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Re: [Freeswitch-users] Early media: Sending 180 instead of 183?
by Peter Olsson 06 Aug '12

06 Aug '12

If you don't want to send 183, don't try to send early media. It's the only way. In the dialplan you posted, remove the line: <action application="playback" data="/tmp/test.wav"/> /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Dennis Skickat: den 6 augusti 2012 12:58 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Early media: Sending 180 instead of 183? we are doing a lot with ISDN-connections, while ISDN in germany seems to be very different from the ISDN in the usa. our carriers asked us to only send a 180 instead of the 183 because, as i wrote above, 183 messages are not mapped reliable to ISUP as written in the specification ITU-T Rec. Q.1912.5. the a-leg sends an IAM or INVITE, but does not receive an answer - this can differ from a-leg to a-leg. we know, that this problem is quite special, but we have the same problems with different carriers. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:501fa15e32761376919998!

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Anyone using FS with Polycom and BLF (speed dials)?
by Yehavi Bourvine 06 Aug '12

06 Aug '12

Hi, We have various Polycom phones which monitor other extensions. Most of them use speed dials/BLF (dialog in SIP level) and some use the "buddies" (presence at SIP level). The first method is preferable as the monitoring phone can differentiate between the state of the line (mainly ringing as opossed to busy). In recent versions (after FS-2983) we fonud the following problem: When a user puts a cal lon hold, dials another call, its state on the BLF moitoring phone is stuck in BUSY. The buddies are ok. Has anyone else seen this problem? Or are we the only ones to use BLF on Polycom... I tried it on various 3.3 trains, and all behave the same. BTW, on SNOM it is ok. Thanks, __Yehavi:

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UniMRCP content type broken
by Ben Langfeld 06 Aug '12

06 Aug '12

I'm testing FreeSWITCH with Voxeo PRISM's media server, via UniMRCP. >From the FreeSWITCH side, it looks like everything works, but the media server complains about an unknown content type, because contrary to its logs, freeswitch appears to be sending application\sssml+xml (too many 's'). Logs here: https://gist.github.com/477e53db58318125cc2c Does anyone have this working, or know how to fix it? Regards, Ben Langfeld

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google voice / XMPP questions!
by Trel 06 Aug '12

06 Aug '12

I am using gvoice per the config and directions on the wiki: http://wiki.freeswitch.org/wiki/Google_Voice We're using a mixed bag of hardware SIP capable phones, linksys and cisco mostly. flow is: google voice <> ipcop firewall / NAT <> dmz'ed freeswitch box(running current) <> bunch of SIP phones the freeswitch server is a dell poweredge, couple xeons at 3.something, four gig of ram, couple sata2 hard drives in raid1. load stays under 14% basically all the time. iostat shows very low usage for basically everything, our pipe is comcast, 20/8. we have, seemingly randomly, audio quality problems. most phone calls are crisp and clear, and one in ten, (Sometimes in streaks of 3-4 calls) we'll have choppy sound both directions. "Hello my name is mike and i'm calling for" sounds like: "llo ame is and alling" this seems to be per-call, while one person in the office is having poor call quality most others are fine at the same time. i have enabled QoS, and prioritized traffic to and from the freeswitch box. our pipe is not saturated during the problems. i've played with VAD, send-silence, and rtp-timer. i've tried changing the codec the phones use with no real effect, i've tried changing the codec used in jingle but we're unable to dial out using anything but PCMU. Where should i look next? Thanks in advance <3 Trel Nadal

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Fax (RX): Missing name/header variable in socket!?
by Dennis 06 Aug '12

06 Aug '12

hi, we are receiving many faxes (RX), but we are missing a variable in the socket. we do get the "remote station id", which is the fax-number itself, but we need the name/header of the incoming fax (for example: COMPANYNAME INC.). thanks for your help dennis

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RTMP Client not responding in Chrome
by Jack 06 Aug '12

06 Aug '12

Is anyone else experiencing problems with RTMP FLEX client in Chrome? Just today it started making chrome become unresponsive. If I allow it to wait about 3 minutes, sometimes it finally starts working. I am experiencing this on my site as well as the conference.freeswitch .org/conf site. It is working fine in IE Safari, and Firefox. jack

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Simultating a modem transaction...
by Tony BECQ 06 Aug '12

06 Aug '12

Hello everybody, Here is my issue. I need to simulate a modem transaction in C. It's a V.21 modem in 200 bits/s. This modem dial a number and waits for an "request to send" sent by the reciever. This is where I have to simulate a modem. The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32 alternations of two tones : 1650 Hz and 1850 Hz, 15 ms each. In my idea, streamFile() function seams to be the good mean. The real issue is that when those tones are sent, the transmeter will send a "message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop bits. 200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0". So, I need to get those 16 digits messages from tones back into numbers. So I need a way to listen the signals and analyse them and this is where I don't know how to do it. Maybe, I have to create a module into freeswitch and here also I need an help to do it. Thanks for your attention. 8hector8

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Re: [Freeswitch-users] tel uri
by B.Tietz＠pinguin.ag 06 Aug '12

06 Aug '12

Hi, as mentioned from IRC: to quote mod_sofia: SWITCH_LOG_ERROR, "URL Error! tel: uri's not supported at this time if there is someone who has an idea, just write :-) regards, Benjamin >Hi, >is there a way to send a tel uri instead of a sip uri in an initial INVITE!? >so I need: > >FROM: <tel:+441234567> >and not > >FROM: <sip:00441234567@domain.sip>

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