May 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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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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Ast/FS Rosetta Stone
by Michael Collins 25 Jun '08

25 Jun '08

FYI, I create a new wiki page: http://wiki.freeswitch.org/wiki/Rosetta_stone I thought maybe those who know about Asterisk and FreeSWITCH both could start dropping stuff in there. I think that people coming from Asterisk might have an easier time if they had some sort of frame of reference. Note that I put a disclaimer at the top - there isn't really a one-to-one Ast-to-FS translation for most of the stuff, but if you said to an experienced Asterisk user that "conf/dialplan/default.xml" is analogous to "/etc/asterisk/extensions.conf" then he/she would at least have a starting point. I'm hoping to add more stuff like the fact that there aren't really "priorities" in FS but that you can use scripting langs, ivr menus, etc. to mimic a lot of that kind of functionality. Anyway, your thoughts and suggestions are welcomed. I will do my best to add stuff as I am able... -MC

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clustered freeswitch
by Vieri 24 Jun '08

24 Jun '08

Hi. I'm completely new to freeswitch and would like to migrate from asterisk. I have a couple of questions though: 1) I understand I won't be able to use my B410P Digium ISDN (mISDN) BRI card yet, right? 2) my TE120P Digium PRI card should work. 3) can freeswitch work in a master-master cluster? Example: FreeSWITCH-1 --- MySQL nbcluster tables --- FreeSWITCH-2 I suppose I can do this if I enable ODBC in FreeSWITCH. But will both FreeSWITCH boxes get "confused" if they use the "same" clustered database (active/active)? (by the way, does the built-in sqlite support clustered tables?) SIP softphones and ATA phones would register via DNS SRV (or custom method) to any one of the servers but only one (the idea is to load balance registrations). Say, extension 101 registers to FS-1 and 102 registers to FS-2. Could 101 contact 102 simply by looking it up the clustered database (eg. querying the table may yield something like sip:102@FS-2)? This would be similar to Asterisk 1.4+ Realtime SIP registrations. In Asterisk 1.2 I use IAX2 friend trunks between the 2 servers and DUNDi for the lookup. Would queues and ring groups also work in a "clustered" freeswitch system? In case of "queues", where is the caller position stored? Who's counting? FS-1 or FS-2? Could the counter be stored in the clustered SQL table so that both FreeSWITCH servers get it right in case calls come in on both servers? example: PSTN---PRI---FreeSWITCH-1 --- clustered SQL --- FreeSWITCH-2---PRI---PSTN Calls coming in from the PSTN reach both FS-1 and FS-2. I would like these calls to enter a "clustered" queue which bridges calls to operator extensions which can be registered on either server. The caller positions should be coherent across both servers. Has anyone done this already? Thanks for your feedback. Vieri

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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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Freeswitch Ldap Integration
by Faraz R. Khan 02 Jun '08

02 Jun '08

First of all- Amazing project. Tired of asterisk deadlocking all the time we have been deploying asterisk with OpenSER as the registrar. Freeswitch is a huge relief! This is an extremely important feature we have been looking for. Asterisk realtime ldap integration is very flaky. I found this page: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_Directory But there are no clues as to the schema, the queries generated and what exactly it can hold in Ldap. I am also curious to know whether sofia's sip registrations, gateways etc can be kept in LDAP. We are basically developing an extensive plugin based control panel and a Asterisk module is already ready. However, we are writing asterisk .conf files for managing asterisk. We would be quite pleased to develop a FreeSwitch Ldap plugin to manage users,sip gateways, groups, features, etc. Though the XML configuration file is extremely easy to parse and write, pure LDAP integration would be amazing. Any pointers on this would be appreciated. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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error sip 500
by Pieter Eduard 02 Jun '08

02 Jun '08

Hi all, Im trying to bridge 2 softswitch (sysmaster and mera mvts 2) using fs, i already setup the dialplan and external gateway in the sip profile but i'm having this trouble when trying to send call to the mvts 2, sometime the call could establish fine and the next minute the call to the same number failed with this error : variable_proto_specific_hangup_cause: [sip:500] variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] variable_hangup_cause: [NORMAL_TEMPORARY_FAILURE] what can cause this error? every failed call is never logged at the mvts2 box as like fs never sends the call but at the fs log i see that it already send the call to the right gateway (mvts2). -Pieter-

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