November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Goip GSM Gateway works great with FreeSwitch!
by Dissident 12 Sep '14

12 Sep '14

http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-… Hi, I've configured this GSM gateway with Freeswitch and it works great. I'm posting this so that if somebody tries to configure this Gateway won't have to spend as much time as a I did :) This is the configuration: *Gateway* *freeswitch/conf/sip_profiles/external/goip.xml* *Tips to check that everything is OK - Remember to remove the PIN number from your SIM card. Double check your SIM card with a Phone. It should not ask you for a PIN. Disable it! Period :) - *sofia status profile internal* (the Goip extension user must appear, I chose ext. 1000 for it...) - * sofia status* (to check that the Goip gateway is properly configured. - On the Goip status page the Phone Status and your GSM Status must be LOGIN - On the Goip status page GSM operator GSM signal must show something. If it doesn't it's because it has not logged properly with your GSM Sim card. Double check it with a Phone. - If the Goip is logged properly with FS you can use your favorite SIP client to log directly into GOIP that is to say. if your GOIP IP is 192.168.2.2 you can log into 1000(a)192.168.2.2 *(again this is not your FS IP is the GOIP IP)* and make an outbound call to an ordinary cell phone. This is useful in case you have trouble with the outbound call. * Things to find out :?? - First I used X-Lite, Windows XP and miniSip server 20 clients to test this gateway. It's kind of strange that with miniSip every time I call to the extension I chose for Goip a tone appears an then I can dial a cell phone number. FS says that this extension is not available, it shows up with the sofia status profile internal though. *I've used with Linux Ubuntu: - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500) - Jitsy Java Sip Client - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client - 2 linksys spa941 - 1 Goip GSM GateWay http://goip.com.ua/download/goip-series-manual.pdf http://goip.com.ua/download/goip-series-manual.pdf http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… I hope you find this info useful, if you have any comments or requests or corrections... please let me know... best regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 5

PD: SIP -> XMPP calls
by Muszalski Michał - Korpo TP 12 Aug '14

12 Aug '14

> Hello, > > We are trying to make some kind of a gateway between SIP and XMPP domain. > We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is > registered as a component in OF). Calls from XMPP to SIP are working fine. > The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing > tone, a the callee (XMPP) has a 'connecting...' message after answer the > call. > Do you have any experience in making calls between SIP and XMPP users? Is > it possible to establish the video calls also? > > Best Regards, > > Michał Muszalski > Senior Specialist > > Orange Labs Poland > Obrzeżna 7, 02-691 Warsaw >

4 4

Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

4 8

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.

10 15

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

2 2

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

5 4

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

4 4

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

3 3

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

5 6

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

6 11