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Any way to get the Dialplan: logging and similar entries to be date prefixed?
by Nathan Neulinger 14 Apr '14

14 Apr '14

Is there any way to get these log entries timestamp prefixed like the rest of the log entries in freeswitch.log? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Snom 370 and Day/Night led
by Mr.Gab 14 Apr '14

14 Apr '14

Hello everybody, I have a FS server with a couple of SNOM 370. Users can manage tha day/night mode by 2 extensions that control a variable. I need to turn on the led near the "night mode button" when the variable is set to "night" and turn off it when the variable is set to "day". Can anyone suggest me how to implement this feature? Thank you in advice crazygabry

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major openssl vulnerability.
by Michael Jerris 14 Apr '14

14 Apr '14

More information available at http://heartbleed.com/ . You should probably upgrade openssl to at least 1.0.1g and re-generate all keys and invalidate old keys.

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Re: [Freeswitch-users] How to check the channel variables "fax_result_code" in the script? such as php script?
by sparklezou 14 Apr '14

14 Apr '14

Hi All, I also checked the wiki https://wiki.freeswitch.org/wiki/Channel_Variables Is there any way to get the channel variables in PHP script? Then it will be very easy to handle the following steps according to the result code. Thanks in advance! 2014-04-14 sparklezou 发件人："sparklezou"<sparklezou(a)163.com> 发送时间：2014-04-11 10:39 主题：How to check the "fax_result_code" in the script? such as php script? 收件人："freeswitch-users"<freeswitch-users(a)lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev(a)lists.freeswitch.org> 抄送： Hi Buddies, Reviewed the wiki, https://wiki.freeswitch.org/wiki/Mod_spandsp How to check the "fax_result_code" in the php script? Also other result info, such as fax_result_text, then could provide more detail response info to the client. Thanks! 2014-03-03 sparklezou

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Random Idea
by Blake Priddy 14 Apr '14

14 Apr '14

You know what would be cool. Is if FreeSWITCH could send out emergent alerts to all end points. That's let's say what a weather radio would get. If there is a emergency alert in your area you can select a group of extensions that would be able to get that NOAA broadcast. Do like a page or something.

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FreeSWITCH Master Branch Week in Review is moving from Monday to Tuesday
by Kathleen King 14 Apr '14

14 Apr '14

Expect the Week in Review to now be sent out on Tuesday afternoons. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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Snom 370 and Day/Night led
by crazygabry 13 Apr '14

13 Apr '14

Hello everybody, I have a FS server with a couple of SNOM 370. Users can manage tha day/night mode by 2 extensions that control a variable. I need to turn on the led near the "night mode button" when the variable is set to "night" and turn off it when the variable is set to "day". Can anyone suggest me how to implement this feature? Thank you in advice crazygabry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Snom-370-and-Day-Night-led-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Audio degrades ON TRANSFER with specific MOH file
by Sean Devoy 12 Apr '14

12 Apr '14

Hi All, I have a user that has claimed that if she picks up incoming calls, the audio is excellent. However, if someone else picks up the call and transfers it to her, the audio is "terrible". While investigating I found that they are using their own custom MOH file. I suspect that the AUDIO configuration is being modified by the MOH feature and leaving her transferred call in a degraded state. But, I don't understand enough to actually see if that is the case. Please review the pastebin file: http://pastebin.freeswitch.org/22302 I think I have trimmed it down to the relevant data. Ext 302 has answered and is just initiating the transfer. If I am correct in my hypothesis, what is the correct solution? Thanks in advance, Sean

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Polycom 4.0.4 is not Heartbleed affected
by Brian West 12 Apr '14

12 Apr '14

Just checked it, Thought you guys would like this news -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Different codec negotiation behaviour
by Torsten Hantzsche 11 Apr '14

11 Apr '14

Hi, i have noticed a difference in the codec negotiation between version 1.2.stable and current master and would like to ask if this is intentional or a bug. My FS codec string in vars.xml contains this: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=opus,speex,silk,G722"/> The UAC (CSIPsimple) sends the following SDP: v=0 o=- 3606049718 3606049718 IN IP4 10.0.0.30 s=pjmedia c=IN IP4 10.0.0.30 t=0 0 m=audio 4000 RTP/AVP 106 99 9 8 3 101 c=IN IP4 10.0.0.30 a=rtpmap:106 speex/16000 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:4001 IN IP4 10.0.0.30 FS 1.2 accepts SPEEX with 16kHz: [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:106:16000:20:0]/[opus:116:48000:20:0] [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:106:16000:20:0]/[SPEEX:99:8000:20:24600] [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:106:16000:20:0]/[G722:9:8000:20:64000] [DEBUG] sofia_glue.c:5355 Substituting codec SPEEX@20i@16000h [DEBUG] sofia_glue.c:3190 Set Codec sofia/internal/ally.mcbeal(a)petitenet.prv SPEEX/16000 20 ms 320 samples 42200 bits while FS 1.5 seems only to check for SPEEX with 8kHz and then jumps to G722 as closest match: [DEBUG] switch_core_media.c:3389 Audio Codec Compare [speex:106:16000:20:0]/[opus:116:48000:20:0] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [speex:106:16000:20:0]/[SPEEX:99:8000:20:24600] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [speex:106:16000:20:0]/[SILK:117:8000:20:20000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [speex:106:16000:20:0]/[G722:9:8000:20:64000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [SILK:99:24000:20:0]/[opus:116:48000:20:0] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [SILK:99:24000:20:0]/[SPEEX:99:8000:20:24600] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [SILK:99:24000:20:0]/[SILK:117:8000:20:20000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [G722:9:8000:20:64000]/[opus:116:48000:20:0] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [G722:9:8000:20:64000]/[SPEEX:99:8000:20:24600] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [G722:9:8000:20:64000]/[SILK:117:8000:20:20000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] [DEBUG] switch_core_media.c:3443 Audio Codec Compare [G722:9:8000:20:64000] ++++ is saved as a match [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMA:8:8000:20:64000]/[opus:116:48000:20:0] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:8000:20:24600] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMA:8:8000:20:64000]/[SILK:117:8000:20:20000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [GSM:3:8000:20:13200]/[opus:116:48000:20:0] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [GSM:3:8000:20:13200]/[SPEEX:99:8000:20:24600] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [GSM:3:8000:20:13200]/[SILK:117:8000:20:20000] [DEBUG] switch_core_media.c:3389 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] [DEBUG] switch_core_media.c:3315 Set telephone-event payload to 101 [DEBUG] switch_core_media.c:2343 Set Codec sofia/internal/ally.mcbeal(a)petitenet.prv G722/8000 20 ms 160 samples 64000 bits The same occurs with SILK too. Is this the new behaviour in FS 1.5 ? Does now the codec string always have to be exactly defined to work for SPEEX or SILK like "speex@16000h,silk@24000h" ? Torsten

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