April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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freeswitch as a session border controller
by Robert McNaught 30 Mar '09

30 Mar '09

Hi, I have been reading the recent news articles about freeswitch. I am particularly interested in using it to replace a session border controller we have (kagoor voiceflow 1000). Has anyone actually used this in production as a session border controller for NAT traversal? Thanks Robert McNaught

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Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 09 Sep '08

09 Sep '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Reg: SS7
by Gopal krishnan 21 Jun '08

21 Jun '08

Hi, Is freeswitch compatible with SS7 technology. How can i Implement SS7 in freeswitch -- Thank you with regards, Gopal,

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Profile configuration and aliases
by Jonas Gauffin 19 Jun '08

19 Jun '08

Hello Question 1: I got the following config: Name Type Data State ================================================================================================= outbound profile sip:mod_sofia@MY.PUBLIC.IP:5080 RUNNING (0) nat profile sip:mod_sofia@MY.PUBLIC.IP:5060 RUNNING (0) default profile sip:mod_sofia@192.168.1.124:5070 RUNNING (0) mydomain.com alias default ALIASED ================================================================================================= 3 profiles 1 alias I can't user "originate sofia/default/auser 8888", I get "Cannot locate registered user auser@default". It works if I use "originate sofia/mydomain.com/auser 8888". Shouldn't both work? -------------------------- Question 2: Isn't this valid? <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="sip_profile=mydomain.com"/> If not, please change the description: "sip_profile: Must be a domain name if you are being a registry server; otherwise can be any string." It produces the following config: Name Type Data State ================================================================================================= outbound profile sip:mod_sofia@MY.PUBLIC.IP:5080 RUNNING (0) nat profile sip:mod_sofia@MY.PUBLIC.IP:5060 RUNNING (0) default profile sip:mod_sofia@192.168.1.124:5070 RUNNING (0) 192.168.1.124 alias default ALIASED ================================================================================================= 3 profiles 1 alias User's cant register using mydomain.com as domain. I tried by adding $${sip_profile} as an alias for the default profile, but that didn't help. -------------------------------- Regards, Jonas

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SRTP in PhonerLite and Freeswitch
by Krzysiek 28 May '08

28 May '08

Hi I have 2 softphones PhonerLite (they support SRTP via SDES ) and the freeswitch (windows RC1 version) server and I wanted to make secure call between those two endpoints (SRTP). I spend whole day on testing this scenario and my conclusions are: - when the option: <action application="export" data="sip_secure_media=true"/> is uncommented, and both enpoints have enabled SRTP then: 1) Initiator of the session sends SIP Invite with a=crypto paramter and supported codecs 2) Freeswitch receives SIP Invite and sends SIP Invite to the receiver (also with the crypto) 3) Receiver receives the SIP Invite with the a=crypto parameter and he sends back supported codecs with 200 OK message (but without a=crypto parametr. Is that ok? I'm afraid not) 4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0 RTP/AVP 19 and no a= parameters! 5) Final result is that the second leg of the session between Freeswitch and receiver has SRTP transport enbaled and the first leg (initiator- Freeswitch) doesn't hear anything - no codecs! However Freeswitch is sending RTP (not SRTP) pacekets to the initiator. Could someone explain to me, what is going on, and why freeswitch doesn't forward codecs accepted by the receiver to the initiator? Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested SRTP with the PhonerLite softphone or any other free softphone with srtp support? When I uncommented: <param name="Inbound-no-media" value="true"> everything works fine. The parameter <action application="export" data="sip_secure_media=true"/> doesn't change anything then (but i cound miss something). Thanks for help Chris

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XML RPC
by Cavalera Claudio Luigi 16 May '08

16 May '08

Hello! I've installed latest svn trunk on a dedicatet machine, so vmware issues are gone. Music on hold now sounds pretty good! :-) I would like to issue some commands to freeswitch via web browser. I think that XML-RPC is the way to go but I've just found this documentation http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC which is not enough for me at this point. Is there any XML-RPC quickstart as the Javascript quickstart? I would like to try some click to dial from a web browser, maybe with a bit of javascript too :-) Thanks a lot! Ciao, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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mod_portaudio send 3 rtp packet/60ms instead of 1 packet/20ms
by Zelei Csaba 14 May '08

14 May '08

Dear all, I tried to use FS in client mode, starting calls with mod_portaudio to our providers gateway ( a Cirpack softswitch ) I experienced that there is 2-3 sec delay in the call, its choppy and robot like. I tested it with a softphone, and an ip phone and everything was fine. I traced back the problem to mod_portaudio sending 3 rtp packet in 60ms instead of 1 packet/20ms. Here is an rtp statistic from a call: (see http://pastebin.freeswitch.org/4307 for the complete list and sip trace) Packet Sequence Delta (ms) 42 26138 0.00 43 26139 0.02 46 26140 45.69 47 26141 0.02 48 26142 2.96 52 26143 56.31 53 26144 5.75 54 26145 0.02 58 26146 51.99 59 26147 0.03 60 26148 2.96 63 26149 42.95 65 26150 17.06 66 26151 0.02 67 26152 2.90 71 26153 56.99 72 26154 0.03 73 26155 0.02 Did anyone else experience similar problems? Is this the desired behaviour, because portaudio get data in 60ms interval or can I set it to 20ms somehow? Thanks, Csaba Zelei

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IP authentication
by Jed Stafford 01 May '08

01 May '08

This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious. I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy. I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required. Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc. -Jed

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Wiki-meet Agenda results for April 30th 2008
by Brian West 30 Apr '08

30 Apr '08

The meeting was great lasted a little over an hour and we covered many things we need to get done. This weeks agenda we went over and documented here: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_04_30 (Please add to or update anything you feel like working on) Please add anything to next weeks agenda that we need to go over here: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_07 The call for the agenda will go out on Mondays for the Wednesday meetings. Please feel free to join in at any time everyone is welcome to participate. If you need to call me your default FreeSWITCH install has a way to call the phone sitting on my desk via 1000*543 (I'm in CST) Thanks, Brian West FreeSWITCH sip:brian@freeswitch.org #freeswitch-docs on irc.freenode.net

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