April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Anyone got ZRTP MiTM working?
by Peter Villeneuve 11 Apr '14

11 Apr '14

Hi, I've asked this question several times before but unfortunately I have yet to get any reply. Has anyone managed to get FS to recognize and enroll a ZRTP enabled client like CSipSimple? For some reason, on my default install from git, when dialing 9787 the lua script answers the call but tells me that it is only available for ZRTP enabled endpoints. Obviously it isn't recognizing the ZRTP hash even though I do see it in the FS logs. I also tried the same experiment with Jitsi and got the same results, even though ZRTP was clearly enabled. So, any clue as to why FS doesn't seem to recognize the ZRTP hash from the clients and enroll as trusted MiTM? Thanks, Peter

3 10

Reading custom SIP headers
by ik 11 Apr '14

11 Apr '14

Hello, I have a SIP Provider that add custom headers to sip . I tried to see them using app info, but they do not appear, but with sofia trace I can see them. For example the provider gives me a header named: Price: 0.25 Gateway: foo The upper case letter is part of the name, and both Price and Gateway are different headers. I tried also to read them using ${variable_sip_Gateway} and ${sip_Gateway}, but the variables are empty. How can I read the extra sip headers ? Thanks, Ido

6 12

About T.38 FAX Machine detail configuraion issue
by sparklezou 11 Apr '14

11 Apr '14

Hi All, I have reviewed the Mod spandsp wiki http://wiki.freeswitch.org/wiki/Mod_spandsp There is an example about T.38 FAX. Samples Here is a working example using FreeSWITCH as a T.38-gateway. I used the following software and devices: FAX_B: Infotec IF2100e ATA_B: Cisco 186 v3.2.1 FAX_A: RICOH FAX2000L (Super G3, 14.400 Baud) ATA_A: Grandstream HandyTone 502 Product model: HT-502 V1.2A Firmware: Program-- 1.0.3.10 Bootloader-- 1.0.0.9 Core-- 1.0.3.6 Base-- 1.0.3.8 FreeSWITCH: git head of 2010-11-19 PRI E1-Stack: openzap Sangoma-Wanpipe: version 3.5.17 TDM-Card: Sangoma A104d My questions: 1. Is there any detail configuration manual for RICOH FAX2000L? I could NOT find on google also. 2. I'm using RICHO MP2000. From the manual, seems it support T.38. But only could set account, NO password. Freeswitch could accept account without password? I also use Wireshark to catch the IP package. seems NO sip regiest info from the RICHO MP2000. Also NO very detail configuration manual from RICHO. 3. Here the T.38 could connect to Freeswitch without ATA or NOT? it's just in LAN? Thanks all! 2014-04-09 sparklezou

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About T.38 FAX Machine detail configuraion issue
by sparklezou 11 Apr '14

11 Apr '14

Hi All, I have reviewed the Mod spandsp wiki http://wiki.freeswitch.org/wiki/Mod_spandsp There is an example about T.38 FAX. Samples Here is a working example using FreeSWITCH as a T.38-gateway. I used the following software and devices: FAX_B: Infotec IF2100e ATA_B: Cisco 186 v3.2.1 FAX_A: RICOH FAX2000L (Super G3, 14.400 Baud) ATA_A: Grandstream HandyTone 502 Product model: HT-502 V1.2A Firmware: Program-- 1.0.3.10 Bootloader-- 1.0.0.9 Core-- 1.0.3.6 Base-- 1.0.3.8 FreeSWITCH: git head of 2010-11-19 PRI E1-Stack: openzap Sangoma-Wanpipe: version 3.5.17 TDM-Card: Sangoma A104d My questions: 1. Is there any detail configuration manual for RICOH FAX2000L? I could NOT find on google also. 2. I'm using RICHO MP2000. From the manual, seems it support T.38. But only could set account, NO password. Freeswitch could accept account without password? I also use Wireshark to catch the IP package. seems NO sip regiest info from the RICHO MP2000. Also NO very detail configuration manual from RICHO. 3. Here the T.38 could connect to Freeswitch without ATA or NOT? it's just in LAN? Thanks all! 2014-04-09 sparklezou

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FreeSWICH training in Shanghai China
by Seven Du 11 Apr '14

11 Apr '14

Hi all, The Chinese community is growing and more and more people are learning and using FreeSWITCH, we will have a 3 days FreeSWITCH training from April 19th in Shanghai China, Obviously the training course will be in Chinese (Mandarin). Welcome to join us. Check the link for details: http://www.freeswitch.org.cn/2014/03/24/freeswitch-peixun-shanghai.html http://fstrainingsh.vasee.com/ Thanks. -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

1 0

Freeswitch+AD
by Nick Zaitsev 11 Apr '14

11 Apr '14

Good day to you. Please Help. How can I do user auth via Active directory? Best regards Nick Zaitsev e-mail nzaitsevc(a)gmail.com

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