May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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e911
by Joli Martinez 17 Jul '14

17 Jul '14

Hello I am looking for an open source e911 service where I can manage multiple locations centrally. Does anyone have any experience with any software that can do this? Thanks,

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IPV6 addr change killed sofia profile internal
by Mario G 21 Jun '14

21 Jun '14

I have a mix of native IPV4/6. A mac runs FreeSWITCH using a static IP4 address and has IPV6 addresses. For some reason (I am not an IPV6 expert) the IPV6 address changes once or twice a day. Everything on the mac runs fine except FreeSWITCH stops all the profiles then restarts them, it then loses the internal profile. The sequence is below, If anyone knows of a FS param or something I can use to workaround this please let me know. BTW, I am not using IPV6 to the external ITSPs, but the Yealink phones have it on, I could turn it off but was hoping for a real solution. Thanks, Mario G 1 - Log shows: [INFO] mod_sofia.c:5286 IP change detected []->[] [2601:9….. 2 - FreeSWITCH stops all 7 external profiles and the internal profile. 3 - All profiles are restarted, but only the external ones start successfully. The internal does not start because the port has not been freed by OS X yet. And FreeSWITCH never attempts to restart it again. I am sure this is related to Jira http://jira.freeswitch.org:8080/browse/FS-5223 but is a bigger problem since I have to turn off IPV6 to stop it. 4 - I have to "sofia profile internal start” to get it going again. PS: Would love to find a way to get FS to keep trying to restart internal rather than give up, would help both problems.

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No Ringback on few calls
by Nagalenoj H. 18 Jun '14

18 Jun '14

Hi, I am facing an inconsistent behavior in getting Ringback for calls. When I call to few numbers, I am not getting ringback tone. I did capture the packets and checked the difference between the call which I get RBT and failed case. I couldn't figure out what is stopping Freeswitch to provide '183 session progress' to the application after receiving '180 ringing' from the server on the other side. I checked the other server and it is sending the tone to FS, it is in FS machine the tone it not forwarded due to some reason. In Freeswitch log, there is no log for "Sending early media" in case of failure case. I've attached the packet capture for the calls. It has both working and non-working case. Please help me in debugging this issue.

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scalability of mod_conference and mod_shout
by martin.hoole＠emailn.de 12 Jun '14

12 Jun '14

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FreeSWITCH - MySQL
by Joel White 09 Jun '14

09 Jun '14

Just a question. I want to attempt to use MySQL as the DB for FreeSWITCH, many of the other systems use MySQL and I would like to keep the data in the same place. The question is, what character set and collation should I use? The FreeSWITCH site explicitly states to NOT use utf8 Thank you in advance

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