June 2014 - 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.

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Skype is likely to stop supporting desktop API by Dec. 2013
by Henry Huang 12 Sep '14

12 Sep '14

Here is the link: http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated I don't know if this mean that we'll no longer be able to interconnect with Skype anymore. And I just want to share the news with this community. Henry

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Params to FS Native.Api() changed. How to get handle of CoreSession?
by Srini K 10 Sep '14

10 Sep '14

Hi, In older version of FreeSWITCH I have been using FreeSWITCH.Native.Api(). For eg, string uuid = new FreeSWITCH.Native.Api().ExecuteString("create_uuid"); activeCalls = new FreeSWITCH.Native.Api().ExecuteString("show calls") I updated FS to latest version. Now Native.Api expects CoreSession as param. FreeSWITCH.Native.Api(FreeSWITCH.Native.CoreSession) How to get handle of CoreSession. Iam using mod_managed. Thanks in advance. Srini

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mod_curl setting content type header
by Jeff Stark 04 Sep '14

04 Sep '14

Hi, Is there a way to set the Content-Type header when using mod_curl in a Lua script? Using curl on the command line I can do the following: curl -H "Content-Type: application/json" -H "Accept: application/json" -X POST -d '{"field1":"val1","field2":"val2"}' http://mydomain.com/ And in a Lua script I can do the following: local my_json = "{\"field1\":\"val1\",\"field2\":\"val2\"}" api = freeswitch.API() response_data = api:execute("curl", "http://mydomain.com/ post" .. my_json) But I don't see a way in the freeswitch API to set the Content-Type header in the above script. Appreciate any help or insight that can be offered. Thanks. Jeff

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Inband DTMF Woes
by dotnetdub 02 Sep '14

02 Sep '14

Hi List, We are using a carrier who insists on sending DTMF INBAND. I was using the start_dtmf application in my dialplan but this was generating a lot of talk off. I've switched to spandsp_start_dtmf application and this seems to work without any issue.! My next issue is that we have a customer who is using asterisk and an IVR that ends up in the voicemail application. Freeswitch seems to be stripping the 'BEEP' out of the RTP that the voicemail application sends. It does it a bit randomly - and I initally thought it was the spandsp start dtmf application. I've been playing with the thresholds and twists etc but it doesn't improve. I've captured the RTP and the leg from the asterisk PBX has the voicemail 'BEEP' but the leg to the carrier doesn't. I've done /event plain all when the server isn't busy to see if I can spot something happening when the BEEP is in the RTP but nothing comes up. It doesn't seem to be the span dsp dtmf stuff causing this. I'm at a bit of a loss. Anyone any ideas? Thanks, Brian

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Not able to send SIP chat
by Juan Antonio Ibañez Santorum 28 Aug '14

28 Aug '14

Hello, I am doing some test to send SIP MESSAGE using chat api without success. SIP domain is 10.3.1.231 Destination reg info is: Call-ID: E950F7906ACA4DC4B2CB5ADEC4C4882D0x0a030191 User: cabinacr1(a)10.3.1.231 Contact: "user" <sip:cabinacr1@10.3.1.145> Agent: SJphone/1.65.377a (SJ Labs) Status: Registered(UDP)(unknown) EXP(2013-08-22 10:23:00) EXPSECS(294) Host: fscentos6 IP: 10.3.1.145 Port: 5060 Auth-User: cabinacr1 Auth-Realm: 10.3.1.231 MWI-Account: cabinacr1(a)10.3.1.231 Command 'chat sip|cabinacr2@10.3.1.231|cabinacr1@10.3.1.231|Hola' returns 'Error! Message Not Sent' Command 'chat sip|sip:cabinacr2@10.3.1.231|sip:cabinacr1@10.3.1.231|Hola' returns 'sofia_presence.c:4309 Not sending message to ourselves!' What is the correct way to send the message? Regards

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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 >

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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>

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Strange media behavior on WebRTC
by Oleg Stolyar 05 Aug '14

05 Aug '14

Hi guys, I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that: 1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference. 2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears. This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both. I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library. Has anyone run into this? Any advice? Here is the relevant excerpt from my dialplan. <condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>

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Limit Database
by Andre 05 Aug '14

05 Aug '14

HI, on Windows I'm trying to use Limit with Microsoft SQL DB but it's still using the sqlight database. I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc setup like this <param name="odbc-dsn" value="sqldb64::" /> I know the odbc works because I have the core database setup the same <param name="odbc-dsn" value="sqldb64::" /> <param name="core-dbtype" value="MSSQL"/> Is there a step that I'm missing? Thanks

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