November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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that the communication was active +CPAS: 4 - hanged up with AT+CHUP - checked with AT+CPAS that the communication was no longer active +CPAS: 0 AT+CFUN indicates a proper registration, AT+CSQ returns a proper signal, and no PIN has been configured - I just don't understand how to get the audio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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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 14 Nov '14

14 Nov '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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XML_Curl and CentOS 7
by Tim Meade 14 Nov '14

14 Nov '14

Good day all.... Playing around with CentOS 7 here and compile accomplished and everything loads/runs. Issue is with XML_CURL. The mod loads, but it never seems to fire. Exact same conf setup on CentOS 6.5 works great. Any known issues or thoughts on this? Thanks Tim

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Set 3 freeswitch to share one single directory.
by Antony Aguirre Morales 14 Nov '14

14 Nov '14

Hi. I have an idea to configure 3 freeswitch but I require that the 3 just consult a directory of extensions, this is possible? Or if there is a way to work simultaneously form two freeswitch directory with a database. better explained in an example: Freeswitch 1 ---- | Freeswitch 2 ---- | ----- Directory of extention Freeswitch 3 ---- |

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How to disable internal profile in freeswitch
by Aqs Younas 14 Nov '14

14 Nov '14

Is there anyway to disable internal profile in freeswitch.? Thanks

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about conference play and bgapi
by 松本祐志 14 Nov '14

14 Nov '14

Hello I have two issues. I am writing in Perl. While 2 people are talking in a conference room, the one person want to play the sound. In "Caller-Username", can you get useless. I have tried the above but, Member: it will not become a *** not found.. my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; I want to play the above in the background It can not play in the next program. my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav); $ con-> bgapi ($ api_cmd); Best regards

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ringing in two devices configurate one extencion.
by antonyam 14 Nov '14

14 Nov '14

Hi, My question was related when i have an extensión and i configure the same extensión in two softphones. When this extensión is reciving a call, one extensión is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ringing-in-two-devices-config… Sent from the freeswitch-users mailing list archive at Nabble.com.

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SIP trunking with Nexmo
by Manish Talwar 14 Nov '14

14 Nov '14

Hi,? I have? make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external?" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name ?Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia@192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4@sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: <document type="xml/freeswitch-httapi"><params></params><work><execute application="set" data="sip_h_api_key=b9c280dd" /><execute application="set" data="sip_h_api_secret=7678b8c4" /><execute application="set" data="sip_h_to=919818753995" /><execute application="set" data="sip_h_from=18188535351 <sip:b9c280dd@sip.nexmo.com>" /><execute application="set" data="sip_h_answer_url=http://services.qpayi.com:8080/ivr/assets/NexmoTransfer.vxml" /><dial name="exten" action="http://localhost:8080/ivr/fsdialmenu/dialendresponse/" caller-id-name="HTTAPI Test" caller-id-number="18188535351" context="default" Dialplan="XML" >919818753995</dial></work></document> here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED? Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) ?as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar?

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