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Freeswitch db for voicemail
by Alex Lake 17 Nov '14

17 Nov '14

Where does one configure the database that the voicemail system uses? Doesn't seem to be using the odbc/mysql database that I specified for core, but I'm not sure if voicemail is supposed to be part of core?

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fs_cli hangs
by akhil garg 17 Nov '14

17 Nov '14

Hi, I am using 1.5.4b freeswitch release. I am running freeswitch by commands "freeswtich" or "freeswtich -nc" running fs_cli hangs. it gives no output. I could see the socket is open by running command "netstat -anlp | grep 8021" "tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 3084/freeswitch" Please help -- regards, akhil

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DTMF strange problems
by Kevin Mathy 17 Nov '14

17 Nov '14

Hi, I'm facing a "strange" problem with DTMF, on a FS server I'm using (Version 1.5.8b git fe2a4d6 2014-02-14 00:03:11Z 64bit). For some calls going through FS, when customers send DTMF tones (using RFC2833), I see those logs : 2014-11-14 15:47:48.516849 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 1:742 > 2014-11-14 15:47:48.516849 [DEBUG] switch_channel.c:489 RECV DTMF 1:742 > 2014-11-14 15:47:49.056857 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF C:640 > 2014-11-14 15:47:49.056857 [DEBUG] switch_channel.c:489 RECV DTMF C:640 > 2014-11-14 15:47:49.356848 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 2:661 > 2014-11-14 15:47:49.356848 [DEBUG] switch_channel.c:489 RECV DTMF 2:661 > 2014-11-14 15:47:49.696846 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 3:665 > 2014-11-14 15:47:49.696846 [DEBUG] switch_channel.c:489 RECV DTMF 3:665 > 2014-11-14 15:47:50.256872 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 4:745 > 2014-11-14 15:47:50.256872 [DEBUG] switch_channel.c:489 RECV DTMF 4:745 > 2014-11-14 15:47:50.796855 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 5:741 > 2014-11-14 15:47:50.796855 [DEBUG] switch_channel.c:489 RECV DTMF 5:741 > 2014-11-14 15:47:51.236850 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 6:664 > 2014-11-14 15:47:51.236850 [DEBUG] switch_channel.c:489 RECV DTMF 6:664 > Here, the customer tried to send "123456". So, I don't know why this "C" character has been sent, and I'm sure the customer didn't send this, because there's no "C" touch on his phone's keypad... And this "C" results in a double "1" received on the IVR server (hosted on another server than FS). (Ex : 123456 => 1C23456 => 1123456) It doesn't happen everytime, but for around 80% of calls... So it's really a problem for us... I looked at PCAPs for concerned calls, and saw that DTMF tones are only sent with RFC2833, not with SIP INFO nor InBand. I first thought that digits may be received simultaneously with RFC2833 and InBand but definitely not. Do you think what could be the cause of that ? I don't use "start_dtmf" in my diaplans, I already saw in other ML topics that this may cause double digit problems. Is there a way to ask freeswitch to only handle 0-9*# characters ? Or any idea, I'll be glad to try anything that can help us ! Thanks for help, Kevin *Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP --

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TLS setup failed
by Emrah 17 Nov '14

17 Nov '14

Hi list, I am able to use FS with SSLv23 with Blink Pro on Mac OS, but that’s about it. I get the following error if I connect with any other device (Bria iOS, Yealink phone, Join Softphone): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) I came across this thread: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-td7587… but it doesn’t seem to apply to my scenario. I am using a commercial certificate. My devices connect to a domain which has an SRV record which points to itself on the SSL port. SSL host is an A record and matches the CN in the certificate. Server cA check is even turned off on certain phones. The only error I get on my Yealink phone is this: Nov 14 11:04:03 SIP [524]: SDL <6+info > [000] SSL_connect (read done) Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL ERROR Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL_connect error I would appreciate to know how I could debug this further. Or if you have any clue at what may be going on. Thanks! Emrah

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Re: [Freeswitch-users] unable to do (raj)
by Kalyani Kulkarni 17 Nov '14

17 Nov '14

> > > originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 > async full) > try 192.168.1.94:9050 async full, the colon is missing in your syntax

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Get Billy Club now on iTunes
by Ken Rice 15 Nov '14

15 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/14orplP Get Billy Club now on iTunes I posted last week about this from my phone but it didn’t work very well. CURSE YOU MOBILE BROWSERS! Anyway, My kid was in a movie 3 years ago when he was 12 and not even old enough to see it but it finally was released and is available on iTunes. He’s a mean baseball player who makes an ordinary kid grow up to be a killer. Check out Billy Club! Only $5 to rent and $10 to Own. http://ift.tt/14orplT

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Hack for Zoiper speex codec payload problem
by Mateusz Bartczak 15 Nov '14

15 Nov '14

Hi All, I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution. Maybe these days some of you can share some hints how to hack this Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec. I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure And yes, I know that zoiper is broken, please don't paste RFC here :) It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken

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Configuring a HUAWEI EM770W for gsmopen
by Charles Devereaux 14 Nov '14

14 Nov '14

Hello I'm new to freeswitch, and I would like to try and use a HUAWEI EM770W with gsmopen. The module is unlocked, and voice is enabled. Voice dialing has been tested and works : using MobilePartner on a windows XP running on virtualbox, I can dial a number, and hear the audio. However, from linux I just don't understand where the voice flux can be obtained or how to configure gsmopen. From http://wiki.freeswitch.org/wiki/GSMopen#Configuration_File I thought it would be some ttyUSB device, but I saw nothing. Here's what I did: - disconnected the usb device from virtualbox menu (so that it would be properly initialized with the at commands, etc) - started minicom, typed atd number followed by a semicolon (for voice mode) - monitored /dev/ttyUSB[1-5] : only ttyUSB2 showed some activity : ^CONF:1 ^CONN:1,0 ^CEND:1,69,29,16 MODE:3,3 - checked with AT+CPAS that the communication was active +CPAS: 4 - hanged up with AT+CHUP - checked with AT+CPAS that the communication was no longer active +CPAS: 0 AT+CFUN indicates a proper registration, AT+CSQ returns a proper signal, and no PIN has been configured - I just don't understand how to get the audio or how to configure May I please ask for some help? I will be happy to update the wiki explaining how to use a EM770W (they're cheap!) if it can be made to work. (Also, I have a problem compiling on a debian sid - first error being in bootstrap.sh: automake is not found while 1.14.1 is installed and in the $PATH) Here are the current settings : AT&V &C: 1; &D: 2; &F: 0; E: 1; L: 0; M: 0; Q: 0; V: 1; X: 1; Z: 0; S0: 0; S2: 43; S3: 13; S4: 10; S5: 8; S6: 2; S7: 50; S8: 2; S9: 6; S10: 14; S11: 95; S30: 0; S103: 1; S104: 1; +FCLASS: 0; +ICF: 3,3; +IFC: 2,2; +IPR: 115200; +DR: 0; +DS: 0,0,2048,6; +WS46: 12; +CBST: 0,0,1; +CRLP: (61,61,48,6,0),(61,61,48,6,1),(240,240,52,6,2); +CV120: 1,1,1,0,0,0; +CHSN: 0,0,0,0; +CSSN: 1,1; +CREG: 1; +CGREG: 1; +CFUN:; +CSCS: "IRA"; +CSTA: 129; +CR: 0; +CRC: 0; +CMEE: 1; +CGDCONT: (1,"IP","mm) ; +CGDSCONT: ; +CGTFT: ; +CGEQREQ: (1,2,0,0,0,0,2,0,"0E0","0E0",3,0,0),(2,2,0,0,0,) ; +CGEQMIN: ; +CGQREQ: ; +CGQMIN: ; ; +CGEREP: 0,0; +CGCLASS: "A"; +CGSMS: 3; +CSMS: 0; +CMGF: 0; +CSAS: 0; +CRES: 0; +CSCA: "+13123149810",145; +CSMP: ,,0,0; +CSDH: 0; +CSCB: 0,"",""; +FDD: 0; +FAR: 0; +FCL: 0; +FIT: 0,0; +ES: ,,; +ESA: 0,,,,0,0,255,; +CMOD: 0; +CVHU: 1; ; +CPIN: ��,��; +CMEC: 0,0,0; +CKPD: 1,1; +CIND: 0,1,1,1,0,0,1,0; +CMER: 0,0,0,0,0; +CGATT: 1; +CGACT: 0; +CPBS: "ON"; +CPMS: "SM","SM","SM"; +CNMI: 2,1,2,2,0; +CMMS: 2; +FTS: 0; +FRS: 0; +FTH: 3; +FRH: 3; +FTM: 96; +FRM: 96; +CCUG: 0,0,0; +COPS: 1,2,""; +CUSD: 0; +CAOC: 1; +CCWA: 1; +CCLK: ""; +CLVL: 4; +CMUT: 0; +CPOL: 0,2,"",0,0,0; +CPLS: 0; +CTZR: 0; +CTZU: 0; +CLIP: 1; +COLP: 0; +CDIP: 0; +CLIR: 0; ^PORTSEL: 0; ^CPIN: ��,��; ^ATRECORD: 0; ^FREQLOCK: 13723540,0; ^GLASTERR: 1; ^CVOICE: 0; ^DDSETEX: 0; ^CMSR: 0; ; ^AUTHDATA: 1,0,"",""; ^CRPN: 0,""; ^WPDST: 1; ^WPDOM: 0; ^WPDFR: 65536,1; ^WPQOS: 255,50; ^WNICT: 0; ; Thanks Charles

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Doc-Sprint Starting Now
by Ítalo Rossi 14 Nov '14

14 Nov '14

Hello all, We're starting the FreeSWITCH doc sprint right now! If you want to help please join #freeswitch-docs and we'll be available to give you instructions. -- Ítalo Rossi

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bridge failover and failure_causes
by Vladimir Getmanshchuk 14 Nov '14

14 Nov '14

For the scenario when used multiply dialstring: <action application="bridge" data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/> failure_causes which was sed to USER_BUSY, only works when received 486 (USER BUSY) after 18x.... If it received 486 just after 100 Trying, it trying to bridge to the next gateway and ignore failure_causes=USER_BUSY... Is it normal behavior? -- Yours sincerely, Vladimir Getmanshchuk

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