June 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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Khushil Dep invites you to their network on Ecademy
by khushil.dep＠gmail.com 21 Sep '08

21 Sep '08

Connect with me on Ecademy Hi I've added you as one of my contacts on Ecademy where successful business people grow their business. Click here to connect with me: http://www.ecademy.com/account.php?op=signup&xref=310185 Thank you, Khushil Dep khushil.dep(a)gmail.com This email was sent to freeswitch-users(a)lists.freeswitch.org To avoid receiving these emails in the future send an email from that address to invite-unsubscribe(a)ecademy.com or go to http://www.ecademy.com/blockinvites.php?e=freeswitch-users@lists.freeswitch… Email:support@ecademy.com Company Registration:3651083 VAT:718 0377 36

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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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Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration
by Diego Viola 23 Jul '08

23 Jul '08

Hi, I'm sorry for my reaction... I know FreeSWITCH is better compared to Asterisk, and I will learn it, with XML or not. I also don't think XML is a bad thing, it's an open standard and I think I will get used to it. @Ognjen, when you say: "You can't be serious that Asterisk/Digium is ok when you ask them for a feature or anything at all? Tony offered them to rewrite obvious broken core to make it better, they said no. I tried 1.4 because of the new features advertised, and I got no sleep beacuse of it for two weeks. What's the point of polishing something that is without any doubt broken in core? I don't understand why are you even looking at FS as an option, if Asterisk is ok for your needs?" What do you mean with "broken core" exactly? I know asterisk has problems but I would like to know what core problems you refer there. I'm just curious about it. I'm looking at FS as an alternative to Asterisk because I been having a lot of problems with Asterisk, and I want a system that works better, the problems I had are the following: http://bugs.digium.com/view.php?id=12628 http://bugs.digium.com/view.php?id=12653 http://bugs.digium.com/view.php?id=12810 And I had more than that... and I keep having problems, I want to move on and have a system that I can trust. Anyway... I will learn FS and XML, and sorry for my reaction... I know Brian is a good and cool guy and he is the guy that introduced me to FS. Peace out, Diego On Sun, Jun 29, 2008 at 10:20 PM, Michael Collins <mcollins(a)fcnetwork.com> wrote: >> This XML argument has gone on since FreeSwitch was unveiled and quite >> frankly I was one of the people railing on Tony for that decision, but > as >> I >> have continued to look at the core and develop soluitons on this > platform >> it >> has become more and more obvious that I was wrong initially... (wow >> someone >> archive tha.. Its the only time you'll see it on a mailing list :P ) >> K > > > So noted! It would be nice to get an audio recording of you groveling > for Tony's forgiveness... we could add it to the demo IVR: > > "...To hear Ken Rice admit the folly of his ways, please press ..." > > :) > > -MC > > P.S. - Can you record that in UWB mode? =D >

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[Fwd: Openser + FreeSwitch Integration.]
by Aadilkhan Maniyar 17 Jul '08

17 Jul '08

Hi All, I would like to try and integrate OpenSER and FS. In this setup OpenSER would be the proxy and the Registrar. The registrations and subscriber info are stored in MySQL database. I would like to use FS as a B2BUA, so that it maintains all the dialog information, different call legs etc. (Something similar to Openser and Asterisk integration) Now my queries: 1. How can I proceed to achieve the above? 2. How can I share the MySQL database so that FS can access the registrations stored by OpenSER. 3. I understand that FS stores its data in SQLite database files. How can I make FS talk to MySQL database. Also I have tried integrating mod_spidermonkey_odbc (http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc) But still the registrations are being stored in the SQLite file and not in the MySQL database. Any help regarding this would be appreciated. Thanks in Advance. Regards, Aadil

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Compile and install mod_cdr
by Rodrigo P. Telles 11 Jul '08

11 Jul '08

Hi Folks, I'm trying to compile and install mod_cdr following the wiki page http://wiki.freeswitch.org/wiki/Mod_cdr but it seems obsolete and broken. BTW I try do add "event_handlers/mod_cdr" on modules.conf and it builds fine. But when I try to run freeswitch it complains with: 2008-05-09 11:35:03 [CRIT] switch_loadable_module.c:727 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_cdr.so **/usr/local/freeswitch/mod/mod_cdr.so: undefined symbol: _ZN12CDRContainer7add_cdrEP19switch_core_session** Does some one here build mod_cdr sucessfully? If so, please share the steps with us. Thanks in advance, Rodrigo Telles

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polyphase lowpass filter
by e schmidbauer 30 Jun '08

30 Jun '08

can someone direct me to the line in the mod_shout.c that enables polyphase lowpass filter? i'd like to enable it when sending to an icecast stream. thanks.

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How to Pass the CLID
by Klaus Teller 30 Jun '08

30 Jun '08

Hi Guys, Forget about the YAML guy and continue with the impressive work. I got one question: when I bridge a leg A, to a number B. How do I pass the number of the caller A to the B. That is, what do I need to do to have B see the number of A on the display? Right now, despite setting the caller_id_num variable (using both sesstion.setVariable, and session.set), I still see a very strange number (area code 690) on B's display. Is the problem lying with the termination VOIP provider? Klaus L'Ivrogne. -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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Sphinx integration
by imthiyaz＠peopletech.co.in 30 Jun '08

30 Jun '08

Hi I am tring to get some opensource voice recognization and i found Sphinx and Sphinx latest versions supports only 16KHz samples.( which asterisk doesn't http://www.voip-info.org/wiki-Sphinx) I planing to test freeswitch ( i think it supports )with Sphinx , any inputs and support information will be appreciated. Thank you Imthiyaz Original Message: ----------------- From: Michael Collins mcollins(a)fcnetwork.com Date: Sun, 29 Jun 2008 19:46:55 -0700 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] [Freeswitch-dev] YAML support asanalternative of XML for configuration > You said: "If XML is the single reason you won't be using FreeSWITCH > then by all means stick with Asterisk, because we aren't going to beg > you to use FreeSWITCH." > > This is sad, you claim that FreeSWITCH is more advanced in terms of > features and innovation than Asterisk, but all I see with your > attitude is that it's more limited and closed-minded. > Please don't be so quick to judge Brian by this response. Or better yet, please don't be so quick to judge FreeSWITCH by Brian's response. The simple fact of the matter is that FS *is* more advanced than Asterisk, regardless of whether or not the devs are willing or able to swap out XML for YAML. > I'm afraid that I will have to stick with Asterisk, at least they > aren't arrogant and closed-minded when you ask for a feature, all I I wonder what the asterisk devs would say if you asked them to switch the conf files out for YAML files? Let's just say that their response would be something similar to: "Uh... no." > want is a more effective way (for me) to configure FreeSWITCH, this is Actually, technically, there's nothing preventing someone from writing a script/module/program that would convert YAML to XML, and even doing it on the fly. Can you honestly say that can be done with Asterisk? Even if it could be done it would be much more difficult than doing it with FS. FS has hooks built into it that allow for this kind of flexibility. Tony knew from the get go that not everyone would be willing/able to hand edit all the XML files. So an important decision was made early: make the XML be both human- and machine-readable. > not going to hurt you or steal your freedom, or change the way you do > things... it's only going to allow me to do things in the way I'm used > to and I like. > > This is exactly why I love Free Software & Open Source, because it > gives me the freedom to do things exactly the way I want, but I see I > wont change your mind... so, good luck with FS. I don't know if changing Brian's mind is actually relevant anyway. The best thing to do would be to write a parser that converts XML to YAML and YAML to XML and demonstrate the coolness/value/etc. of a YAML-based configuration. If YAML is indeed as great as you feel it is then I'm certain that there will be others who would agree and would be willing to support the effort to make a YAML-based config module/parser/thingy. So let me rephrase Brian's last words: If the XML-based nature of FreeSWITCH is the *only* thing that is preventing you from using it then please consider the possibility that you might be missing out on something very cool and very powerful. You aren't forced to use XML, but it is the easiest of the alternatives, at least at the start. Once you get hooked on FS you'll realize that it's a better choice for a switching platform. The choice certainly is yours, and no one here will begrudge you your right to choose. My parting advice: choose wisely. -MC > > Diego > > On Thu, Jun 26, 2008 at 11:15 PM, Brian West <brian(a)freeswitch.org> wrote: > > Diego, > > I can already tell you Its not going to happen. The choice was > made > > early on and it was the correct one. It's not something that can be > > changed or modular. If XML is the single reason you won't be using > > FreeSWITCH then by all means stick with Asterisk, because we aren't > > going to beg you to use FreeSWITCH. > > > > Read this for more info http://www.freeswitch.org/node/123 > > > > /b > > > > On Jun 26, 2008, at 9:14 PM, Diego Viola wrote: > > > >> Hi everyone - > >> > >> I really love this project... I love the design you guys took with FS, > >> I love the stability, the features, the openness of it, it's the VoIP > >> platform I always dreamed of it... FreeSWITCH is what Asterisk should > >> have been. > >> > >> But I just don't like one thing: XML. > >> > >> I also don't intend to create a flame war or troll with this thread, > >> please take this as a suggestion of someone that really wants to use > >> FS... XML is the only reason why I still stay with Asterisk, I just > >> can't like XML, and I don't like Asterisk as well, I want FreeSWITCH > >> stability, features, etc, but I can't like XML, not even if I try to, > >> it's too slow to use/write, verbose and complex. > >> > >> As a sysadmin point of view, I prefer parameter=value but I realize > >> the advantages of XML (data serialization, etc), this is also why I > >> don't use mod_dialplan_asterisk, because I think it's time for > >> something better than INI files, but XML isn't for me, I'm a > >> Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want > >> something simple to use, read and write, and I think YAML is the best > >> of the both worlds. > >> > >> I would like to have more than one option to configure FreeSWITCH, > >> also, not everyone likes to do the same thing the same way and it > >> would be really cool if FS gives this freedom to people... > >> > >> I would really like to see YAML as a configuration option in > >> FreeSWITCH, give me this any day and I swear I don't toch Asterisk > >> _ever again_, YAML can be used for data serialization and it's more > >> simple than XML, Ruby projects like Rails uses it. > >> > >> Take a look here: > >> > >> http://yaml.org/spec/1.2/ > >> http://yaml.org/ > >> > >> "YAML(tm) (rhymes with "camel") is a human-friendly, cross language, > >> Unicode based data serialization language designed around the common > >> native data types of agile programming languages. It is broadly useful > >> for programming needs ranging from configuration files to Internet > >> messaging to object persistence to data auditing. Together with the > >> Unicode standard for characters, this specification provides all the > >> information necessary to understand YAML Version 1.2 and to create > >> programs that process YAML information. " > >> > >> Please consider it, I'm also willing to pay for a mod_yaml, other than > >> this I really love FreeSWITCH and I think it's great, I will switch in > >> a heartbeat if you guys add support for it ;-) > >> > >> Best regards, > >> > >> Diego > >> > >> _______________________________________________ > >> 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 > > > > > > _______________________________________________ > > Freeswitch-dev mailing list > > Freeswitch-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > _______________________________________________ > 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 _______________________________________________ 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 -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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