Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: Hi its good to hear any compare document between Vicidial and this project No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
On Mon, Dec 21, 2009 at 09:48:44AM -0800, Michael Collins wrote: On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson and...@hijacked.us wrote: On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: Hi its good to hear any compare document between Vicidial and this project No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Now *that* is a feature worth paying for! ;) Also, I thought you had a community edition vs. a professional edition? If so could you explain the difference? I've managed to avoid that thus far, I suspect that something like an outbound campaign manager (which could be implemented as just another media type) might be something to fall under that sort of split, but right now the release includes everything we've got (including an integration module that's probably of limited use to anyone else - but its a good example of how to build your own). Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Interfacing to RabbitMQ
On Sun, Dec 20, 2009 at 07:38:27PM +0330, afshin afzali wrote: Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. You could try to use mod_erlang_event and the erlang rabbitmq client (in native message passing mode). I've never worked with rabbitMQ however, I just know a little about it. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version = R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our own CRM - spicecsm. The URL supports interpolation for variables like callerid, clientid, call type, etc. The supervisor view is a little hard to describe via static images, but you're able to drag and drop agents into another profile (empty profiles are hidden when not dragging an agent), drag agents onto an agent to send them the call, and there's also various right click menus available. Oh, and I forgot to mention this before; this system is in 'live testing' and the goal is to do a final deployment sometime in January. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Building on Windows
On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but.., I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as Unsupported, although the Wiki says that you only need VC++2005. What does unsupported mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like make current and make sounds may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Route Non-Call Data to Agent Through Queue
On Thu, Dec 10, 2009 at 10:26:39AM -0800, Shaun Clark wrote: I have an application where I would like to route both calls and other requests through the same queue to the same agents, for example the same agent might take a call and then right after that take a chat. But, the chat server we use is separate from our phone system. What I would like to do is basically route some text, i.e. new chat chat_id_goes_here through to the agent. Is this possible with FreeSwitch? The idea being the soft-phone would receive this text and we would write code to catch this message do the appropriate action on our CRM. Thanks! I did this by writing my own external queueing (in erlang) and simply parking the calls in FS and adding them to my external queue (along with emails, voicemails, etc). With asterisk I added a fake call to the queue with some channel variables that referenced the external data I was really putting in the queue and I listened for the 'BRIDGE' event on the AMI and sent the agent the external data then. I'm not sure mod_fifo needs to be a universal queue - but maybe you could do what you want via api_after_bridge and uuid_chat or something crazy? You'd have to script whatever soft-phone you're using to be smart about that though. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Eavesdrop error?
On 12/2/2009 9:19 PM, Lars Zeb wrote: Is this reasonable given it was the only call in FreeSwitch at the time? How can this situation be corrected in the future? As a workaround, you can eavesdrop with 779, and use * to navigate channels. -- Andrew Thompson ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Holiday routing examples
Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS missing in action (was: How to run IVR application)
On 11/27/2009 12:59 AM, ovvenkat wrote: Hi MC, I have created won sample application yesterday, It was working fine. Today, I checked that my local ip has changed. so, I changed the domain(IP) name in sip-account settings in my x-lite configuration. After that x-lite is not able to register with FS. I am getting error like Registration error 405 : Method not Allowed . Could you please tell me ,why its happening ? Wait, what? First, don't re-use an existing thread, messages have a tendancy to get ignored/lost that way. Did the IP of your FS change, or of your PC? I would expect local ip to mean your DHCP'ed address from your Internet connection. That should have no bearing on the IP of your FS. Go find your FS and make sure *IT* is still on the IP you expect it to be on. If your PC running x-lite is also your FS, you may have other issues with IP address changing that I don't know how to handle, as I've only used static IPs for FS. (Or, try connecting via localhost, 127.0.0.1 instead, until you're ready to really start using FS.) -- Andrew Thompson ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re-routing calls to PSTN
On 11/26/2009 6:02 AM, Otis wrote: Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Since I don't have a hard line, I do something like: include extension name=2800 condition field=destination_number expression=^2800$ action application=bridge data=sofia/gateway/YOURPROVIDER/18005551212/ /condition /extension /include -- Andrew Thompson ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Business/holiday hours routing
On Wed, Nov 25, 2009 at 11:21:25AM -0700, Adam Ford wrote: Awesome, thanks Andrew, I will have to keep an eye out for that patch. Here's my patch in its (probably) final form. http://eagle.bsd.st/~andrew/mweek2.diff It includes a usage example that covers all but 2 of the US federal holidays (memorial day is a real toughie). I'm just waiting on Tony to green light it for commit. If the patch looks like a mess in your browser, blame the XML :) Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Business/holiday hours routing
On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: Is there a standard module for FreeSWITCH out there that people use for routing calls based on business hours and a holiday schedule? Or is everyone just creating their own in the XML dialplan?(which seems pretty simple) I can't seem to find anything on the wiki, but might just be searching for the wrong thing. I am relatively new at FreeSWITCH and would rather follow what the community has decided is the best practice, instead of trying to reinvent the wheel myself. I assume you've seen this: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing I have a patch that'll let you specify the nth day of the nth week via wday=3,4 for the 4th tuesday in the month. This willl let you do vacations like thanksgiving, MLK day, etc as well. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Business/holiday hours routing
On Mon, Nov 23, 2009 at 08:17:46PM -0600, Brian West wrote: He's working on it for SVN... I recommended the format and to add the phases of the moon and zodiac signs just for giggles. I'll probably get a patch in this week (or early next) I'm thinking of changing the format so that week of month becomes its own value so you could compare against mweek as well as wday so thanksgiving + extension becomes something like condition mweek=4 wday=5-6 month=11/ If I really get ambitious I'd also like to allow wday=mon-fri so I don't always forget that days are 1-indexed from sunday :) Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accessing Config Info From Database
On Fri, Nov 13, 2009 at 11:28:55PM +0100, Leon de Rooij wrote: Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration Or, if you're really crazy, the erlang module can do it too (even dynamically): http://wiki.freeswitch.org/wiki/Mod_erlang_event#XML_search_bindings :P Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] playback from hadoop
On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? mod_shell_stream is undocumented, but from reading the code I gather it works like this: Module calls fork() and in the child process it runs an arbitrary shell command (specified in its config file?). The parent process then reads raw audio data from the child process and uses it as an audio source. So basicially you could write the shell command in anything, so long as it outputs raw audio to FS. Or maybe I read the code wrong when I skimmed over it. If you do get it working, please contribute some documentation to the wiki. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] playback from hadoop
On Wed, Nov 11, 2009 at 11:02:10AM +0800, mark morreny wrote: Hi Sorry to ask again. I know the command to copy file from hadoop file system to somewhere else. But how do I make a shell command to output raw audio? What command is it like? Is it like play()? I am confused. I was very nice and wrote up some documentation (and 2 examples) on the wiki page at http://wiki.freeswitch.org/wiki/Mod_shell_stream Now you know everything I know about using this module (which is a very cool module, by the way - thanks Tony). Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] playback from hadoop
On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: Hi, Does anyone know how to playback based on files from hadoop storage. There is a libhdcp, and java api. Is there anyway to put together a sample middle piece to move files from hadoop to freeswitch using memory space, so there is no disk I/O? Any feedback or suggestion will be greatly appreciated. mod_shell_stream might work, if you can just spit out the raw audio to the shell. Or write another stream module that works with libhdcp. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] portaudio error
On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: Hello Debian lenny with svn15321 freeswi...@internal load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswi...@internal 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** Try installing the alsa development headers, it's got some stupid name on debian like libasound2-devel or something. Then re-build the portaudio module and library (a couple well placed make cleans should do it). Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking
On Fri, Oct 09, 2009 at 11:01:34AM -0700, Michael Collins wrote: On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker r...@rupa.com wrote: Oh, definitely. I think both use cases (existing, and park in next avail slot and read back slot #) are useful. They do have some usefulness. The most important reason for this feature, though, is for the person who handles calls all day and moves them around. That person will most likely have a pretty good idea of which parking stalls are available. However, I do like the idea of a webby interface showing what stalls are in use... mod_snom and the programmable LEDs would be a handy way to do it too. I implemented a poor-man's version of a key system using that, but I think I'll replace the parking portion with the valet parking stuff when I get a chance. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skill-based ACD
On Thu, Sep 24, 2009 at 08:40:16AM -0700, msc wrote: On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman r.klooster...@mtel.nlwrote: Hello Michael, Do you still want to follow up on this? I?m having difficulty gathering the old stuff in an understandable form. Also, it looks like the open source ACD Spice Telephony by Andrew Thompson can do just what you might need. I had totally forgotten about Andrew's stuff! Unless people want to build their own 100% community/free/DIY version of a skill-based ACD then I say let's all play with SpiceCSM and help improve it. I'd certainly appreciate the feedback (and the kick in the ass to improve some of the rough spots and documentation). Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skill-based ACD
On Sun, Sep 20, 2009 at 08:42:40PM +0200, Remko Kloosterman wrote: This actually sounds very good Andrew. You even have an agent interface. Do you have plans for a outbound campaign dialer? I know of a commercial dialer that is good in it's predictive algotithm, but very bad when it comes to campaign management. I don't have plans for an 'autodialer' in the traditional sense but I do have plans for some sort of campaign dialer - the idea is to use an API to load numbers to be called into a queue and the agents will just pop those stub calls off the queue and then the system will originate the call to the indicated number. This does mean that you'll be wasting agent time on voicemail/ringouts/whatever but hopefully you'll piss less people off. In addition, then you can farm out the system that decides the numbers to be called and in which order to an external system. An autodialer would certainly be possible under the current system, I just don't really care to implement one. Patches accepted, although really an autodialer might be better off remaining a binary-only module add-on (to prevent the doing of evil becoming too cheap :) ). And yes, to my knowledge it will remain under an open-source license for the forseeable future. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skill-based ACD
On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote: I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC I guess nobody paid any attention to my Cluecon presentation... :( http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based ACD that uses FS for its voice components. I havent pimped it here in quite a while but here's some of its major features * Skill based routing * Priority Queues (instead of just FIFO) * Multiple call types (voice, voicemail and email are currently supported, instant message support (via libpurple) is prototyped) * Outbound call support (no autodialer though) * Distributed system so you can aggregate multiple FS instances/locations into one big 'virtual' callcenter * Web-based agent and administrative interface There's quite a bit more, but that's the overview. The project is finally approaching a 1.0 after over a year of development - I hope to deploy it in production sometime around the end of this year or the beginning of 2010 (replacing my previous custom asterisk solution). You can grab the code at http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or git clone that URL). All you should need to run it is a modern erlang release (R12B5 or newer) and ruby/rake to run the build. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ERLang configuration callbacks
On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: I seem to be missing something in implementing the ERLang callbacks for Freeswitch. Our Freeswitch server is starting and getting registered with ERLang, we're invoking the bind for configuration, but I'm not seeing any of my callbacks fire. What am I missing? The most obvious thing is that you're trying to catch info messages using handle_call. The erlang module doesn't use the OTP protocol for messages so handle_call/cast won't ever fire for messages sent from the freeswitch module. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ERLang configuration callbacks
On Tue, Sep 15, 2009 at 02:29:14PM -0400, Andrew Thompson wrote: On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: Oh wait, I see what you're doing, you catch all the fetch requests in the handle_info and then make a call to another process to get the XML. My question is why are those both in the same process? The handle_info part is fine, but then the pid you make a gen_server:call to *must* be a different process or you'll hit a timeout and the process will exit. However, I can't start 2 copies of that process (one to catch requests, one to return XML) because your module always does a bind! Regardless, I at least get fetch requests when I run your module, I can't make it return something without refactoring it, but it does receive the requests at least. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] I got ERLang to fire a configuration request
On Tue, Sep 15, 2009 at 08:38:41AM -0600, Mark Sobkow wrote:
I still need to stuff the Freeswitch PID into global storage somewhere
so the process that's handling the configuration requests can send the
reply without crashing (it's just getting a node id, not a Pid), but I
seem to be on my way to configuring Freeswitch via ERLang.
Looking at your code, assuming I read it right, you should be able to
just replace:
FreeSWITCHNode ! SomeMsg.
with
{api, FreeSWITCHNode} ! SomeMsg.
The freeswitch module doesn't really have a pid, since it's not a real
erlang node, it's all faked and all messages to a pid or a registered
process on the C node go to the same place.
Andrew
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Re: [Freeswitch-users] FS 1.0.4 erl configure error
On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load Balacing
On Mon, Aug 31, 2009 at 04:56:48PM -0300, Juan Manuel Vicente wrote: Hello: I am using two boxes, with same domain, to register about 1000 users working together a DNS box doing round robin to resolve two differentes Ips one to each box. This two boxes are using the same database (through unixodbc). When I use the funcion sofia_contact or sofia profile status both boxes show the status of the user. Does not matter where the resgistration was did. But when I try to call the user this call fails, the call only works if the call is sent through the same box where the user is registered. I had the same problem except that I was using the multicast module to replicate registrations, I added a sofia config parameter to optionally rewrite the fs_path part of the sip URI on replicated packets so I could hairpin the sip messages through the original registrar. This doesn't help you directly but maybe it'll give you an idea? Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
On Fri, Aug 21, 2009 at 04:15:13PM -0300, Rogelio Perez wrote: Hi Everyone, I'm working on a PBX project for the Sheevaplug ARM based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU. So far I've found a big difference between Freeswitch and Asterisk performance times. This is a comparison of the time it takes them to perform different actions: startup Freeswitch: 3 min. startup Asterisk: 2 sec. call extension Freeswitch:6 sec. call extension Asterisk: 0 sec. shutdown Freeswitch: 6.5 sec shutdown Asterisk:0 sec. reload config Freeswitch: 1 sec. reload config Asterisk: 1 sec. Both were built from sources natively (no cross-compiling), and they use the default startup configurations. I have managed to lower the Freeswitch times by disabling most of the modules and recompiling, but it is still far away from Asterisk (i.e. FS startup time 2.5 min). 1. Is there any way to further improve Freeswitch performance for the ARM architecture? 2. Can this be related to the lack of a FPU (the Sheevalug emulates the floating point operations). 3. On the startup I see this error repeated many times: [ERR] switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related? Try making where freeswitch stores it's sqlite databases /usr/local/freeswitch/db (by default) a ramdisk. I've had this vastly improve FS performance on embedded devices. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Error trying to use PHP ESL
On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote: Hi, I'm trying to get started with the ESL using PHP. I compiled the ESL, then phpmod according to the wiki instructions, but then when I try the examples in the libs/esl/php dir, they fail saying: PHP Fatal error: Cannot redeclare ESLconnection::__construct() in /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132 Checking ESL.php on line 132, I see there are several different declarations for the function __construct() with different parameters each, which makes sense, but doens't work. I am using PHP 5.1.6, is there a required minimum higher than that or something? What could be the problem? Someone in the IRC channel mentioned this too. I looked at it briefly and it looks like the latest 'swigall' screwed it up. The original reporter said he'd file a jira, but you may want to check yourself and if not make one yourself. In the meantime, the previous version of the file was reported to work if you really need it. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] event socket vs erlang
On Tue, Aug 04, 2009 at 07:08:36PM +0800, mark morreny wrote: Hi, I have seen people using both event socket and erlang to control freeSWITCH externally. What is the pros and cons of using event socket vs erlang? It depends on if you want to use erlang or not. The erlang module provides most of the event socket functionality plus a couple extras (dynamic XML bindings ala xml_curl, intelligent message delivery, etc). If you're not already planning to use erlang, it's probably better to dig out the relevant event socket library module for your langague instead. Andrew - author of the erlang module ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Intercom error with SNOM
On Tue, Jul 14, 2009 at 08:53:07AM -0700, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a 401 Unauthorized in the siptrace and a No Matching gateway found in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set Challenge/Response off, Enable intercom on and Type of Intercom Answering to Handsfree. The error is still 401 Unauthorized. What more do I need to do? Try disabling the intercom option and setting the 'answer after policy' to 'only in idle'. I think there's a bug in the snom firmware, see also: http://forum.snom.com/index.php?showtopic=1790st=0gopid=3688#entry3688 Andrew ___ Freeswitch-users mailing list Freeswitch-users@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] mod_snom dialplan demo not working
I'm trying to replace our aging nortel system with a FreeSWITCH based system using snom 3[267]0 phones. I've been doing okay so far but I'm running into a brick wall when I try to run the mod_snom demo in the dialplan. I've setup the 'line 2' function key to type button with the number being 'message' like it says on the snom wiki and when I dial extension 9000 the button lights up, but pressing the button does absolutely nothing. I can't figure out what I'm missing here. Any advice? Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_snom dialplan demo not working
On Thu, Jul 09, 2009 at 08:10:41PM -0500, Brian West wrote: It needs tons of work its not so demo tastic ... Is it incomplete or just there's no documentation on how to do it? From the sip trace it looks like FreeSWITCH is sending the phone the right stuff just when I hit the button that was programmed nothing happens. I want to be able to *use* this feature of the snom phones, not just play with the demo anyway, so if there's some development needed to make it work I'd be happy to take a stab at it. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Originate - using own uuid
On Thu, May 07, 2009 at 01:47:26PM -0700, Simon Tang wrote:
Hi there,
I know there's a feature that allows us to create a uuid and then use
that uuid to do an originate (api create_uuid). Is there any badness if
I don't do the create_uuid and just do the originate using my own
uuid.
I experimented and made the following originate call:
api originate
{origination_caller_id_number=16041234567,originate_timeout=60,originati
on_uuid=abcdefg}sofia/gateway/icall/16041234567 park
...and this works, but just want to make sure if there's any badness of
doing so...that is, if I can guarantee that the origination_uuid I will
be passing in *WILL* be unique.
You also need to be sure that FreeSWITCH isn't going to use it at any
point in the past, present or future ;). It's probably better to use
create_uuid if at all possible.
Andrew
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[Freeswitch-users] [ANN] Spice Telephony 0.3 - an open source FreeSWITCH/Erlang callcenter
Well, it's been a few months since I mentioned this project last here, so here's an update over my last announcement (see http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010048.html ) Things have improved a *lot* since the last time I mentioned it: * Support for inbound calls (somehow last time that wasn't even finished) * Support for outbound calls * Support for ringing agents on a softphone or ringing agents on a POTS line * A web interface for managing the system (managing agents, skills, queues, etc) * A web interface for agents to manage their state (go released, go idle, indicate when they're done with wrapup, etc) Along with a boatload of minor features and even more bugfixes. This release is actually something you can play with without knowing Erlang, we've even included a boot script to setup and run everything for you. So, if you're interested in a distributed, fault tolerant callcenter platform built on top of FreeSWITCH I invite you to check out http://wiki.opencsm.org/wiki/index.php/Spice_Telephony , the rest of the opencsm.org site and especially the other telephony pages on the wiki and, if you're interested, please give it a try and let us know what you think. Just remember that it's still very much a work in progress (although I hope to be able to give a cool demo at ClueCon in August :) ). Andrew Thompson - opencsm.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
On Tue, Feb 24, 2009 at 10:49:24AM +0100, Leon de Rooij wrote: Well, this works, I feel a bit stupid now :-] Now it's time to play with it.. Nah, bad choice of defaults on my part. Defaulting to 0.0.0.0 is much more consistant and compatible. For some reason I was trying to emulate the event socket, not an erlang node. Thanks for finally making me solve the problem instead of just working around it. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
On Mon, Feb 23, 2009 at 11:09:28AM +0100, Leon de Rooij wrote: Everything is running on an Ubuntu Hardy Xen domu with kernel 2.6.24-23-xen. Oh, this might explain some things.. Erlang is version R12B5 and was compiled from source with options -- enable-hipe, --enable-smp-support en --enable-threads. I'm running this too. FS is trunk version 12197. Fine too. I did copy the configuration file to ~freeswitch/conf/autoload_configs Also, I just checked the 'empd -names', after both FS and an erl shell have been started: r...@erlyfs:~# epmd -names epmd: up and running on port 4369 with data: name ldr at port 57114 name freeswitch at port 8031 So that should be fine.. Yes, that's correct. I also tried loading mod_erlang_event from modules.conf, and starting FS as root, but - not surprisingly - that didn't make any difference. I've been looking in wireshark, what exactly is going over the line, and the strange thing is, that erl opens a TCP connection, a SYN packet is sent to FS, after which FS immediately returns an RST/ACK packet and thus closes the connection.. I still don't see anything in the FS CLI. Is there anything I can do to get more verbose output from FS - esp info about why the connection was closed ? It looks like ei_accept_tmo is the one resetting the connection, not my code. I can't even get an error out of it when, for example, I telnet to port 8031, it just closes the connection instantly with no error to the console. Is it possible that something is screwy with the loopback device in a xen guest? Can you get normal erlang nodes on that host to net_adm:ping each other? Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
On Sat, Feb 21, 2009 at 03:42:24PM +0100, Leon de Rooij wrote:
Hi Andrew,
Thanks for your help so far, I hope you can help me a bit further as I
don't get any reply from the FS erlang node, or so it seems.. Here is
what I've done:
- The erlang_event.conf.xml is unchanged:
configuration name=erlang_event.conf description=Erlang Socket
Client
settings
param name=listen-ip value=127.0.0.1/
param name=listen-port value=8031/
param name=nodename value=freeswitch/
param name=cookie value=ClueCon/
param name=shortname value=true/
param name=encoding value=string/
/settings
/configuration
You actually installed this to the right place? It's not installed by
default... The defaults *should* be sane anyway, but I'm just checking.
- mod_erlang_event is not loaded in FS.
- First I start epmd -d -d
epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0
epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369
epmd: Sat Feb 21 13:12:56 2009: starting
epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop
- After that I load mod_erlang_event in FS:
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324
mod_erlang_event_load() sections 16
2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858
switch_loadable_module_load_file() Successfully Loaded
[mod_erlang_event]
2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240
switch_loadable_module_process() Adding Application 'erlang'
2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260
switch_loadable_module_process() Adding API Function 'erlang'
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401
mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426
mod_erlang_event_runtime() Connected and published erlang cnode at
freeswi...@erlyfs
- For which epmd gives the following output:
epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4
epmd: Sat Feb 21 13:13:36 2009: got 25 bytes
* 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65
|..x._h...fre|
* 0010 65 73 77 69 74 63 68 00 00 |
eswitch..|
epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ
epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031
epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1
epmd: Sat Feb 21 13:13:36 2009: got 4 bytes
* 79 00 00 01 |y...|
epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for freeswitch
- Then I start an erl shell on that same machine with erl -sname ldr -
setcookie ClueCon. Output of epmd:
epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5
epmd: Sat Feb 21 13:16:24 2009: got 18 bytes
* 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72
|..x.,M...ldr|
* 0010 00 00 |..|
epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ
epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396
epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5
epmd: Sat Feb 21 13:16:24 2009: got 4 bytes
* 79 00 00 01 |y...|
epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for ldr
As far as I understand the freeswi...@erlyfs node cannot be seen with
nodes() ? So does that mean that I also cannot net_adm:ping() it ?
Yes, it's a 'hidden' node, as all non-erlang nodes are. However, it
should be visible in the output of epmd -names.
Anyway, I tried sending some tuples as is shown on the wiki, but I get
no reply:
(l...@erlyfs)1 {foo, freeswi...@erlyfs} ! {api, status, }, receive X
- X after 1000 - timeout end.
timeout
(l...@erlyfs)2
- Epmd gives some logs:
epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6
epmd: Sat Feb 21 13:19:09 2009: got 13 bytes
* 00 0b 7a 66 72 65 65 73 77 69 74 63 68
|..zfreeswitch|
epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ
epmd: Sat Feb 21 13:19:09 2009: got 23 bytes
* 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 |
w.._h...free|
* 0010 73 77 69 74 63 68 00 |
switch.|
epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for freeswitch
epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6
- And in tcpdump on lo, I see that epmd is contacted after which some
traffic was sent to FS:
13:19:09.535293 IP 172.31.0.13.34678 172.31.0.13.4369: S
2875169966:2875169966(0) win 32792 mss 16396,sackOK,timestamp
17946545 0,nop,wscale 6
...
13:19:09.536834 IP 172.31.0.13.4369 172.31.0.13.34678: . ack 15 win
512 nop,nop,timestamp 17946546 17946546
13:19:09.536923 IP 172.31.0.13.47054 172.31.0.13.8031: S
2868322908:2868322908(0) win 32792 mss 16396,sackOK,timestamp
Re: [Freeswitch-users] mod_erlang_event compile problem
Leon, I can't replicate your issue, at the very least I'd expect you to see the Ignorable error in ei_accept - probable bad client version, bad cookie or bad nodename warning. What OS/Erlang version are you using? Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: Hi, I wanted to try out the mod_erlang_event module. I have Erlang R12B5 compiled and it's in the same location as the Makefile specifies (/usr/ local/lib/erlang/...), but running make in the src/mod/event_handlers/ mod_erlang_event goes wrong: Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so it didn't show up for me. Thanks to MikeJ for the fix suggestion. Also, after this, FS goes haywire after loading the module and spews out these messages continuously: You don't have the erlang port mapper daemon running (epmd). mod_erlang_event needs it to be running in order to be able to register itself as an erlang node. On your system; epmd isn't in $PATH so my system() call that tries to start it fails. I've made the module init system fail properly instead of looping indefinitely as well as print a slightly more helpful error message now. Let me know if you have any better luck :) The fix is in-tree as of r12192. Thanks again for the bug report. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform
On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote: Have you planned any predictive dialer features ( like VICIDIAL )? As Ken Rice mentioned, this isn't really the focus of the project - it's more for inbound and directed outbound (calling campaigns to specific people/businesses - not everyone in the phonebook). Primary focus is inbound (multi brand, skill based routing, dynamic wrapup times, etc). Expect a new release sometime soonish that actually does something useful (accepts and routes inbound calls from FreeSWITCH to an agent). Also; public source control. There's just some additional corporate nonsense that I have to sort out (again) before that can go live. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dynamic Dialplan
On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote: Also depending on what your Timeframe is like there is a distributed queue mechanism with skills based routing on the way... It even managed to route 2 calls in a row this week ;) Still a ways off from anything production grade tho. Andrew ___ Freeswitch-users mailing list Freeswitch-users@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] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH
I'd like to announce the first beta release of a cross-platform ruby/tk GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's not particularly fancy, but I needed a cross platform softphone with good voice quality that was debuggable and didn't have a ton of features to confuse the users. I couldn't find one so we built one. I've got some sparse documentation up at: http://opencsm.org/wiki/index.php/Spice_SoftPhone And you can download it from http://opencsm.org/download . It's under the MPL and I've been cleared to re-licence my other FreeSWITCH related projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris and OSX (it used to work on linux, I assume it still does). Comments/complaints/bugreports welcome. It's definitely still got some rough spots (I don't think it'll run without a controlling terminal, for example), but we're going to be polishing it up and hopefully putting it in production here in the next few weeks to replace a very buggy closed-source phone we've had to endure far too long. Please download it if you're interested, the download count helps us continue working on this kind of stuff :) Andrew - opencsm.org ___ Freeswitch-users mailing list Freeswitch-users@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] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform
Hi, Today I'd like to announce the open-sourcing of a distributed callcenter platform I've been designing/building using FreeSWITCH/Erlang. The goal is to allow multiple callcenter branch offices to operate seamlessly as a whole, or even just scale one large one beyond hardware/software limits by partitioning the inbound lines/agents across multiple servers. You can read more about the design here: http://opencsm.org/wiki/index.php/Spice_Telephony And a quick overview of planned features: * Inbound/Outbound support * Media type agnostic (voice, email, chat, video(?)) * Skill based routing * Agent phone agnostic (SIP or POTS line) * Dynamic wrapup time * Web or fat client interface And you can download a very early work in progress snapshot of the code from http://opencsm.org . We're also open sourcing our customer service management application too, which is also available for download there. If anyone is interested in callcenter development using FreeSWITCH and/or Erlang, we welcome participation at any level. We don't have public source control or a bug tracker yet, but please bear with us; they're on the way. Thanks, Andrew Thompson - opencsm.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
