May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Compiling Freeswitch for Android?
by GillesToo 25 Apr '14

25 Apr '14

Hello Searching the archives only seems to return information on how to use SIPDroid as an SIP client to connect to a Freeswitch server. Since Android is built on Linux, and Freeswitch has been compiled for compact devices, I was wondering if someone had tried compiling it for Android? I could use a small IVR on a smartphone to handle incoming calls. Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Andr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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DID providers - any thoughts? (jan 2013)
by Cal Leeming [Simplicity Media Ltd] 22 Apr '14

22 Apr '14

Hello everyone, I've been recently reviewing several DID providers, and looking for some feedback. I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. The criteria I've been looking for is; * Able to provide either US or UK numbers * Exposes an API that allows control of purchasing and routing * Guarantees premium call quality, good latency, and inbound CLI. * Does NOT operate on a market/exchange basis * Trading as a telecoms provider for more than 5 years * Has a reputable name within the community or industry So far, I have only found the following providers that meet this criteria; * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around £10k in set up fees. Thanks in advance Cal

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avmd from mod managed
by Malay Thakershi 24 Mar '14

24 Mar '14

Hello, Is there a way to do AVMD from mod_managed? I see JavaScript example technique from http://wiki.freeswitch.org/wiki/Mod_avmd but it seems signature of StreamFile is different in mod_managed. In JavaScript, second argument to StreamFile is a function (OnInput) where detection is done while is mod_managed, it is sample start number. So is AVMD possible from mod_managed? Thank you. Malay

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FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Lenny's killing it.
by Gabriel Gunderson 26 Jan '14

26 Jan '14

No way! Lenny is on a roll! It's too perfect (listen in order)! 1) http://itslenny.com/recording.php?file=4d8cde0cc5a4d86ccf221aa5b2903e95 2) http://itslenny.com/recording.php?file=4fdd3f445fff8e1b73456af602367b6e Classic! Gabe

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TLS error?
by Federico Beffa 27 Dec '13

27 Dec '13

Hi, I'm trying to setup FS as a gtalk client. I'm working on Linux Ubuntu 10.04 and have cloned the git FS repository about 1 week ago. I can compile everything with no errors. However, after configuring client.xml and starting FS I see the following error message in the freeswitch.log file: 2010-07-20 23:38:56.392853 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_ dingaling.so **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init** TLS comes from the following Ubuntu packages: libgnutls26, libgnutls-dev. I would appreciate any suggestion on how to investigate further the problem. Could it be a gnutls library version incompatibility? A bug in FS? In the ubuntu gnutls packages? Or simply an error in the configuration? Thanks for you help! Fede

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TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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Client TLS certificate setup
by adahary 26 Dec '13

26 Dec '13

I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well.I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this:tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sipstport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sipsI will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients.Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Gateway Registration Issues
by Aloysius Lloyd 15 Dec '13

15 Dec '13

Hi All, I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the following gateway configuration <include> <gateway name="wwc_test"> <param name="auth-username" value="username"/> <param name="username" value="14161231234" /> <param name="password" value="password" /> <param name="proxy" value="sbc1.iristel.net" /> <param name="realm" value="iristel.net" /> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> The above configuration is not working. I have the following error in cli Error. *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration Failed with status Conflict [409]. failure #1* ==== Also carrier saying there is conflict in the contact field.Here is how they receive now. *Contact: <sip:gw+test_iristel@10.20.30.1:5080 ;transport=udp;gw=test_iristel>.* But the contact field should be like below Contact: <14161231234@10.20.30.1:5080; Any suggestions? Thanks in advance. Lloyd

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FreeSWITCH not sending P-Asserted-Identity
by Vik Killa 29 Nov '13

29 Nov '13

I cannot get FS to send the P-Asserted-Identity header instead it always sends the P-Preferred-Identity header. According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I must set the privacy flag in order for P-Asserted-Identity to be used, so here is my dialplan entry and the SIP INVITE that is sent: <extension name="outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="lcr" data="$1"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="privacy=yes"/> <action application="privacy" data="yes"/> <action application="set" data="sip_h_Privacy=id"/> <action application="bridge" data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> </condition> </extension> ------------------------------------------------------------------------ INVITE sip:+17165555555@YYY.YYY.YYY.YYY:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp Max-Forwards: 68 From: "1000" <sip:7165551001@XX.XX.XX.XX>;tag=8U2rQ1pmXXaDQ To: <sip:+17165555555@YYY.YYY.YYY.YYY:5060> Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d CSeq: 27864844 INVITE Contact: <sip:mod_sofia@XX.XX.XX.XX:5060> User-Agent: FSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 209 X-FS-Support: update_display,send_info P-Preferred-Identity: "1000" <sip:7165551001@XX.XX.XX.XX> v=0 o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 17984 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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