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Weekly News and Notes from the FreeSWITCH Team!
by Ken Rice 12 Aug '14

12 Aug '14

Hey FreeSWITCHers, Last week was a quiet week on the conference bridge as everyone was in Chicago at ClueCon! This week we¹ll be getting back into the swing of things with a few minor changes to the normal conf call schedule. Tuesdays 1400 (2PM) Central (1900 GMT) Weekly Docs Team Call. Join us and help push the FreeSWITCH Docs Project Forward! This week, we¹ll be working more on the migration from Mediawiki to Confluence Wednesdays 1200 (noon) Central (1700 GMT) ClueCon Weekly! - This week we¹ll be doing a little Post Mortem of ClueCon and getting back in to the swing of things! If you missed ClueCon but want to find out what you missed join us then! Thursdays 1400 (2PM) Central (1900 GMT) FreeSWITCH BugHunt and Bug Marshalls Call. Join us to help us sort, classify and trouble shoot open Jiras, along with sorting and testing patches and pull requests for inclusion into the main tree. We¹ll even help you with your open ticket if you join us. Fridays at approx 1PM Easter (Following the VUC Call) will be the Friday Free For All, Join in and chat with other users and developers of the FreeSWITCH Project. Whats the FreeForAll? Its just that a FreeForAll. Topics will vary wildly from FreeSwitch to others. Talk with you all soon! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Timers
by Andre 12 Aug '14

12 Aug '14

HI, can anyone tell me what these timers do and what are the best values to set them at for performance?

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RTP when client is behind symmetric nat
by Kamil Nigmatullin 12 Aug '14

12 Aug '14

I am a bit suprized that there is no questions regarding support clinet's behind symmetric nat? I understand that Sofia handles SIP behind NAT correctly. But what about RTP? Is that supported and ho it is managed? Thank you.

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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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OpenSSL security advisory
by Michael Jerris 11 Aug '14

11 Aug '14

Please note the release of a new OpenSSL earlier today. We recommend you upgrade to the latest release immediately as a number of the addressed issues will effect freeswitch https://www.openssl.org/news/secadv_20140806.txt

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mod_conf : play tmp/uuid.wav!userleft on member exit.
by Stephen Dame 11 Aug '14

11 Aug '14

I've successfully been able to modify dialplan to prompt caller for name, record it and announce in conference per wiki. solution works great. Adding a few lines of code makes sure they can only enter existing conferences. Now that I have their name in temp wav file, trying to decide best approach to play name!" is leaving conference" into conference when they leave. Before heading down a path. Wanted to seek some advice on some possible approaches. 1) Thought I may be able to set exit sound in conf profile with UUID but not sure variables are expanded at run time. 2) The other approach is to monitor ESL conference events and play back the UUID file when its detected that they have left (separate process) 3) Modify source of mod_conf to hardcode playing of /tmp/uuid when member leaves only if the file exists. Is there an easier way I'm missing? Regards, Stephen

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Presence + Transfers
by Chris Tunbridge 11 Aug '14

11 Aug '14

Currently i have Aastra and Cisco having this behavior, working on getting a polycom lab setup. FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit) I have my extensions 101, 102 and 103 So if i call in, and answer on 101, 102 and 103 will successfully show that 101 is on the phone. If i then transfer the call from 101 to 102, 102 and 103 will show 101 get off the phone, however 101, 103 never show that 102 is on the phone. further transferring the call from 102 to 101 or 103 also never shows that they are on the phone *conf/sip_profile/internal.xml:* http://pastebin.freeswitch.com/23020 *conf/autoload_configs/presence_map.conf.xml:* http://pastebin.freeswitch.com/23021 Dial String is as follows: <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@ ${dialed_domain}):transfer_fallback_extension=${dialed_user}}"/> This behavior is similar on Jitsi, but because i set it to do peer mode, it checks every 30 seconds so after 30 seconds it shows that the person (in this case 102) is on the phone.

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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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Ubiquiti UniFi-VoIP and FS
by Kurt Albershardt 11 Aug '14

11 Aug '14

According to http://www.nojitter.com/post/240168789/ubiquiti-networks-enters-the-voip-ma…, Ubiquiti is using FS in its USG http://www.ubnt.com/unifi-switching-routing/usg/ Anyone have any details? thanks~

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Re: [Freeswitch-users] Originating calls
by Michael Jerris 11 Aug '14

11 Aug '14

https://wiki.freeswitch.org/wiki/Loopback On Aug 2, 2014, at 11:19 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello Freeswitch users. > > Currently when I originate a call in my IVR application, I execute the API originate command like this: > api originate {variable1=data,variable2=data}sofia/gateway/outbound/<number> &park() xml default > So yes I have a gateway called outbound and this originates a call by dialling the specified number on that gateway and parking it. I get the UUID and can then do stuff with it. > > But what I really want to do is originate a call by putting the specified phone number through the dialplan so Freeswitch can work out which endpoint to use. I am sure this must be possible but I can't figure out how to do it. > > Just to summarise, I want to originate a call to a number which might be a local extension or it might be a number that is reached through a gateway. Why can't I do something like: > api originate {variable1=data,variable2=data}<number> &park() xml default > I want Freeswitch to work out which Sofia profile and endpoint to use. I want Freeswitch to park the call in the same way as above and give me the UUID. Is this possible? > > Can anyone help with this please? >

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