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Re: [Freeswitch-users] Monitoring Registrations/Users
by Dave Redmore 22 Mar '10

22 Mar '10

Thanks for the reply. I think that for me, the reason that the registrations are getting dropped or calls are getting dropped is because we are running across "best effort:" Internet connections - DSL and Cable - and we are using off-the-shelf ATAs from Grandstream and Linksys. So, stuff happens and we lose calls - but if I could Poll those far ends, I can be more proactive when that happens. If I could send a SIP packet of some sort every minute and make sure I get a response, then I know those devices are ready to receive a call and if they aren't I can proactively find out why. Dave ----- Original Message ----- From: "Michael Collins" <msc(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 22, 2010 5:03:04 PM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] Monitoring Registrations/Users You can monitor in various ways, but a simple way to start out is by using either a script with ESL or just doing: fs_cli -x "sofia status profile internal" and parsing the results to look for whichever SIP registrations are of interest. As far as finding this useful, I'm sure there are some who would find it useful in at least a few cases. However, my guess is that they would be more interested in know why the registrations are getting dropped in the first place... -MC On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore < dave.redmore(a)spigotsystems.com > wrote: Been working with Freeswitch to implement a simple T38 proxy to provide outbound faxing to some customers. Works very well, on initial testing, using the proxy-media option. Nice NAT detection/SBC functionality "out of the box". Generally very easy to get my head around and get it to do what I wanted, One bit of functionality I've been looking for in a softswitch for a while, and have never seen, is to somehow monitor the status of users/registrations and alert somehow if there is a problem. In my situation, we have connected some customers with VoIP phone service for their small offices. Typically, just a simple ATA that gives them 1-4 trunks and a couple of DIDs. The problem has always been that if their ATA drops it's registration or is not available, no one knows it until a customer tells them through some other means that their phones aren't working. This is a huge problem, obviously, and happens more than I would like. It seems like it wouldn't be too difficult to create some sort of monitoring function which can send an Invite packet every little while and make sure the UA responds with a 100/Trying and then disconnect. Basically polling any devices that I deem as needing to always be available and sending an email or something if one isn't. Does this seem useful to anyone else? Has anyone seen anything like this out there? Thanks, Dave Redmore Spigot Networks, Inc. _______________________________________________ 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

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Bridge or Intercept?
by Fraser Redmond 22 Mar '10

22 Mar '10

I want to call an extn, then when they pick up connect them to another existing channel - later I'll run this via event_socket, but for now I'm doing it in javascript (I want to do database updates before and after attempting the connection.) As far as I can see, I have two options, and I'm interested in which is better, and more importantly - how to get either of them working! 1) Create the new session, then intercept the existing channel: var NewSession = new Session('{ignore_early_media=true,hangup_after_bridge=false}user/ 12610(a)domainname.com'); NewSession.execute('intercept', 'cafc3449-9000-47bc-8b28-257748d7ec40'); The newSession phone rings and on pickup both channels get hung up. The ExistingChannel gives a call status in the CLI of [DESTINATION_OUT_OF_ORDER] 2) Create the new session, and create a session object for the existing session and bridge them: var NewSession = new Session('{ignore_early_media=true,hangup_after_bridge=false}user/ 12610(a)domainname.com'); var ExistingSession = new Session('cafc3449-9000-47bc-8b28-257748d7ec40'); bridge(ExistingSession, NewSession); (This has a debug message saying "BRIDGE THREAD DONE", but not much else that seems unusual.) The sound files that are playing in ExistingSession continue, and after about 10sec A notice appears in the CLI indicating a session.hungup event has fired, but both phones remain on the hook and hanging up either doesn't affect the other. Cheers, Fraser

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Monitoring Registrations/Users
by Dave Redmore 22 Mar '10

22 Mar '10

Been working with Freeswitch to implement a simple T38 proxy to provide outbound faxing to some customers. Works very well, on initial testing, using the proxy-media option. Nice NAT detection/SBC functionality "out of the box". Generally very easy to get my head around and get it to do what I wanted, One bit of functionality I've been looking for in a softswitch for a while, and have never seen, is to somehow monitor the status of users/registrations and alert somehow if there is a problem. In my situation, we have connected some customers with VoIP phone service for their small offices. Typically, just a simple ATA that gives them 1-4 trunks and a couple of DIDs. The problem has always been that if their ATA drops it's registration or is not available, no one knows it until a customer tells them through some other means that their phones aren't working. This is a huge problem, obviously, and happens more than I would like. It seems like it wouldn't be too difficult to create some sort of monitoring function which can send an Invite packet every little while and make sure the UA responds with a 100/Trying and then disconnect. Basically polling any devices that I deem as needing to always be available and sending an email or something if one isn't. Does this seem useful to anyone else? Has anyone seen anything like this out there? Thanks, Dave Redmore Spigot Networks, Inc.

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auth incoming calls
by Brian May 22 Mar '10

22 Mar '10

Hello, In debugging that previous problem I noticed another problem. How do I get freeswitch to authenticate incomming calls? As far as I can tell it is not working: 483.970655 192.168.87.13 -> 202.173.153.98 SIP/SDP Request: INVITE sip:9999@202.173.153.98, with session description 483.972989 202.173.153.98 -> 192.168.87.13 SIP Status: 100 Trying 484.180018 202.173.153.98 -> 192.168.87.13 SIP/SDP Status: 200 OK, with session description 484.208932 192.168.87.13 -> 202.173.153.98 SIP Request: ACK sip:9999@202.173.153.98:5060;transport=udp 490.634429 192.168.87.13 -> 202.173.153.98 SIP Request: BYE sip:9999@202.173.153.98:5060;transport=udp 490.637862 202.173.153.98 -> 192.168.87.13 SIP Status: 200 OK Yet the setting is appears to be enabled: <param name="auth-calls" value="true"/> -- Brian May <brian(a)microcomaustralia.com.au>

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

I deactivated iptables, no router, no nat, only 2 public IP, 2 softphones connected on the same FS which is on a public IP to. default config, result -> rtp timeout crazy that's not working since I can make 2 softphones behind a firewall work.... ok that's enough for me tonight ! ;) ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, March 22, 2010 1:55 AM Subject: Re: [Freeswitch-users] rtp timeout yet >I replace public ip by these fake one... > > ----- Original Message ----- > From: "Jason White" <jason(a)jasonjgw.net> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Monday, March 22, 2010 1:49 AM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >> Madovsky <infos(a)madovsky.org> wrote: >>> result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060 >>> >>> 10.10.10.132 = softphone1 registered from public IP >>> >>> 11.11.11.= softphone2 registered from public IP >>> >>> 10.10.10.138 = FS >> >> Those are private addresses; there's a NAT device involved somewhere, and >> that's likely to be where your problems are. >> >> >> _______________________________________________ >> 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

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How to find the number of live calls for a specific dialplan
by sharad 22 Mar '10

22 Mar '10

Hi All If there any way to find out the number of live calls for a specific dialplan. Regards

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sip-force-contact: which to choose?
by Fabio Pietrosanti (naif) 22 Mar '10

22 Mar '10

Hi all, i read in latest source code of FS (of yesterday) that for the handling of sip-force-contact now there is a new option: NDLB-connectile-dysfunction-2.0 Which are the improvement of NDLB-connectile-dysfunction-2.0 respect to NDLB-connectile-dysfunction? I read that the 2.0 also make the users registered with "AUTO-NAT 2.0". Fabio

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Now Hiring for CudaTEL UI Programmer
by Anthony Minessale 22 Mar '10

22 Mar '10

As most people know, the CudaTEL uses FreeSWITCH internally as the telephony engine. We're now looking to expand the development team: We're looking for a UI programmer with jquery experience. See: http://jobs.jsninja.com/job/1598/jqueryjs-ui-developer-at-barracuda-network… Anyone interested, please apply to this forum or email jobs(a)freeswitch.org Thanks. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Re: [Freeswitch-users] Open source softphone that can handle g729
by Emil Ivov 22 Mar '10

22 Mar '10

> sip communicator is crap. Its trying to do way too much and does not > a single thing very well at all. Well, that's a bit vague. Maybe if you've had/have specific issues with it, you could drop us a note so that we could fix them. In any case the project has improved a lot during the last years, and even months so you might want to give it another try one of these days. > If they have G729 its not legal to use. Yes we do have it, and it was implemented as a part of our participation in GSoC 2009. People can use it perfectly legally as long as they buy the corresponding licenses (just as with any other client). Currently use of the codec is disabled by default and whoever needs it has to enable it in the code, after acquiring the necessary license. Cheers, Emil > > /b > > On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote: > >> I believe sip communicator does handle g729. It has a strong >> community and a plugin model (based on osgi) that makes it easily >> (ok, kinda...) customizable. Of course the >> FSComm/givesomemoneytotheproject option is the best one. In any >> case: http://sip-communicator.org/ >> >> Ciao, R >

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Migrating voicemail to new freeswitch instance
by Adam Ford 22 Mar '10

22 Mar '10

How would I migrate people's custom recorded voicemail greetings to a new install of FreeSWITCH? I finally got upgraded from 1.0.4 to SVN trunk. Thanks, -Adam

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