Re: [Freeswitch-users] Q931 TE State Timer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Stefan, On 12.06.2009 19:21, Stefan Knoblich wrote: Umm, you've been doing duplicate work then. :( Well, I implemented just one timer by now. So not much time has been wasted ... The version of ozmod_isdn i have been working on is completely stateful and has a couple of timers already implemented. Very good :) And i remember giving you the location of the git repository on IRC, earlier this year. (But never got any feedback) Sorry, at that time we talked about q931 to pcap. By now I thought state timers are still not done, so there wasn't a reason to test it regarding state timers bt today things changed and I will download your openzap. regards Helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKNfDw4tZeNddg3dwRAkThAJ4iPiZ4ZOAKKWpmKdCbjM8oM5mH6QCeMo5m pV4Y1/hpO7osV8cuInYJd2o= =zoZu -END PGP SIGNATURE- ___ 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] Zoiper reject freeswitch calls
Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. ___ 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] Live Upgrade Techniques
freeswitch-users-boun...@lists.freeswitch.org wrote: OTOH there will be a bit of trouble getting the internal state out of all those modules and libraries... in particular sofia :D We have talked quite some about this, its a major job, easily months of work for multiple programmers. We would love to do it but its not on any roadmaps at this time. Could this be also achieved in hardware via ATCA ? en.wikipedia.org/wiki/Advanced_Telecommunications_Computing_Architecture Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ 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] When to use OpenSIPS/Kamailio in front of FS?
OK, thanks. 2009/6/14 Nik Middleton nik.middle...@noblesolutions.co.uk: Anything that's dedicated undoubtedly has less load that something that's multifunctioned. However the lack of any conversations on front ending a SIP server to FS would likely indicate that no one's found a requirement for it at this time. I would truly hate to see discussions of theoretical performance advantages of one SIP server over another, when in my view, I have yet to reach any real world limit with FS. My FS servers are handling 100,000+ calls/day per server and are probably only at 50% capacity. (I see no point in beating a server to pulp when it's relatively cheap to add another if required) Regards -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Gavin Henry Sent: 14 June 2009 21:34 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ 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 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 -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ 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] When to use OpenSIPS/Kamailio in front of FS?
Well, we currently have a scenario where this seems to be the most logical setup currently. We provide PBX as a Service (SaaS), and want to have a cluster of FreeSwitch servers handling registration and presence. Introducing OpenSIPS in front will allow a couple of features, which I don't see how would be implemented in a good way without anything in front of FS: - true loadbalancing with the loadbalancer module - Live migration of calls to another server to take FS down for maintenance - no need for 100% SRV support in the SIP clients Best regards, Even André On 15. juni. 2009, at 10.47, Gavin Henry wrote: OK, thanks. 2009/6/14 Nik Middleton nik.middle...@noblesolutions.co.uk: Anything that's dedicated undoubtedly has less load that something that's multifunctioned. However the lack of any conversations on front ending a SIP server to FS would likely indicate that no one's found a requirement for it at this time. I would truly hate to see discussions of theoretical performance advantages of one SIP server over another, when in my view, I have yet to reach any real world limit with FS. My FS servers are handling 100,000+ calls/day per server and are probably only at 50% capacity. (I see no point in beating a server to pulp when it's relatively cheap to add another if required) Regards -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Gavin Henry Sent: 14 June 2009 21:34 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ 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 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 -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ 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 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] Is Freeswitch ready for prime time?
On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? Hi Paul - We've used FS over the last 18 months or so to handle millions of calls - some wholesale in/out, some IVR, some calling card, some callthrough - with a total value in the millions of dollars; we have no complaints. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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_php needed
hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to-get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: I couldn’t agree more. We’re working with a group that are developing a massive PHP based music application. They are experts in PHP and MySQL but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate with the FS event socket, allows them to work on the areas they know best and not worry about the telephony side too much. We went the lua route, and don’t use the dial plan at all. My view is to keep all db access and processing out of FS as much as possible. With the event socket you simply don’t need to embed anything apart from the essentials. We are now processing 100,000+ call setups a day (4 hours) per server all using php scripts to drive the application. We may well ultimately use C++ instead of PHP for the event socket comms, but right now PHP does just fine. Regards *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* 13 June 2009 21:57 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] mod_php needed Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net http://irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation
[Freeswitch-users] Segmentation fault with record_session
Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: extension name=skypiax condition field=destination_number expression=^2909/(.*)$ action application=record_session data=/tmp/data.wav/ action application=bridge data=skypiax/ANY/$1/ /condition /extension And here's how I trigger it: *freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA 2909/userBBB* The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: *freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) * Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei ** ___ 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] Asterisks to Freeswitch CALL REJECTED
I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p24031735.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] is there any available gui for freeswitch using cake php?
is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] is there any available gui for freeswitch using cake php?
http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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] is there any available gui for freeswitch using cake php?
Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] funny effect after minimizing xml files
Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d ___ 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] is there any available gui for freeswitch using cake php?
On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? AFAIK, no fully developed GUI available yet, just curious, why are you finding a GUI instead of wikipbx or pfsense? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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] Is Freeswitch ready for prime time?
Paul Mahler p...@ringcarrier.com wrote: I have a large project coming up. I'm interested in using Freeswitch instead of SER and Asterisk. What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? Others have already addressed most of your questions. I would like to point out, however, that the FreeSWITCH developers offer support, by way of a consulting service, on a commercial basis. If you're running FreeSWITCH in a commercial setting and encounter complex issues that require expert advice or attention from developers, consider entering into a consulting contract. This will also help to fund the project and ensure that the development community remains as active as we all want it to be. ___ 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] funny effect after minimizing xml files
Durk de Beer durk.deb...@isp.solcon.nl wrote: Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect I don't understand the problem, but my general advice is this: learn to read the FreeSWITCH logs carefully. Make sure that the log level is set to debug, as it is in the default configuration, then carefully check the log files to see which dialplan extension matched and how the call was processed. ___ 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] funny effect after minimizing xml files
how to help without seeing your dialplan? On Jun 15, 2009, at 6:26 PM, Durk de Beer wrote: Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d ___ 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 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] funny effect after minimizing xml files
You've probably deleted the start/end markers from your dialplan matches..? It might be easier to help if you posted or pastebinned your dialplan. --Dave Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Live Upgrade Techniques
Michael: We are using 5.0, and i think we tested this feature quiet a while
ago and there was no CDR problem.
Raymond: Thanks for hint i'll try it...
On Fri, Jun 12, 2009 at 6:54 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
Well, Nextone for instance has a database the keeps most of the state of
calls, and it’s replicated between the two nodes. (I seem to recall the
database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways,
the CDRs still get truncated when there’s any kind of switchover.
But Nextone is a closed system with limited services. As MikeJ mentioned,
it was discussed for FS, but it’s a LOT of work to get that state
synchronized. And, every custom app/module would have to register and
support recreating their state.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Saeed Ahmed
*Sent:* Friday, June 12, 2009 7:39 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Live Upgrade Techniques
No idea at all,
It’s a commercial SBC.
I wish if we can have same functionality in FS.
- Saeed
--
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Even André
Fiskvik
*Sent:* Friday, June 12, 2009 3:04 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Live Upgrade Techniques
Can you comment some more on how this is configured?
Would it be something that could be added to the wiki in the SBC setup
page?
Best regards,
Even André Fiskvik
On 12. juni. 2009, at 12.16, Saeed Ahmad wrote:
I've experience with a commercial SBC, these are two machines running in
cluster mode. In that case if one SBC is going down then other will take all
new calls including the call which were active on broken SBC (SIP only).
Thats quite ideal for wholesale traffic where the SBC will never be idle.
On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh jo...@defyne.org wrote:
On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to drainstop
isn't that big of a deal unless you're in some sort of hurry, right?
Give it an hour or so to drainstop, then kill 'em.
Yes that's exactly what I'm trying to do. The problem is some people
will
only try one IP address.
Clients that don't properly implement SRV/NAPTR and fail over need to be
smacked. :) (not customers but software that fails to do that)
Yes I'm sure much of their software can do this but it has been set up for
static numeric IPs. And getting the IP changed is a week-long process for
some customers!
Would it not be simpler to try to do something with re-invites or
REFER,
assuming your endpoints support it?
That was actually plan A. I already added a property in sip_profile
called
failover_redirect, which specifies another server to try if FS can't
allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.),
by sending back a SIP 302 Moved Temporarily response, instead of 503 Max
Calls In Progress.
You can't send a 302 to a call thats already established.
Yes and I don't want to touch established calls - those calls can stay
there until they drop. This is sent to new requests when
switch_core_session_request fails in mod_sofia.
Turns out not all my endpoints support it :(
AKA broken endpoints. :)
Some are broken. Some just have this feature disabled. For 'security
reasons'. You know the drill.
{P^/
John
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Re: [Freeswitch-users] Segmentation fault with record_session
Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com wrote: Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: extension name=skypiax condition field=destination_number expression=^2909/(.*)$ action application=record_session data=/tmp/data.wav/ action application=bridge data=skypiax/ANY/$1/ /condition /extension And here's how I trigger it: freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA 2909/userBBB The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei ___ 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 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] Asterisks to Freeswitch CALL REJECTED
Hi, if you understand spanish look at: http://www.freeswitch.es/node/61 Regards ___ 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_php needed
Did you actually use ESL with the php wrapper when you tried? You can do all those things from outbound event socket fairly easily. That mod_php you saw, never worked it was just a stub and it didn't actually ever work when the guy who added it totally disappeared, I removed it from tree. And you can still do event socket over localhost on the same box if you so choose. If you really want a mod_php it's entirely possible but it would probably cost you upwards of 5k in development costs. 2009/6/15 Christian Löschenkohl christian.loeschenk...@xpirio.com hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to-get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: I couldn’t agree more. We’re working with a group that are developing a massive PHP based music application. They are experts in PHP and MySQL but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate with the FS event socket, allows them to work on the areas they know best and not worry about the telephony side too much. We went the lua route, and don’t use the dial plan at all. My view is to keep all db access and processing out of FS as much as possible. With the event socket you simply don’t need to embed anything apart from the essentials. We are now processing 100,000+ call setups a day (4 hours) per server all using php scripts to drive the application. We may well ultimately use C++ instead of PHP for the event socket comms, but right now PHP does just fine. Regards *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* 13 June 2009 21:57 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] mod_php needed Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net http://irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented
Re: [Freeswitch-users] Zoiper reject freeswitch calls
To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. ___ 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 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] Zoiper reject freeswitch calls
Yes, I can reproduce this, FYI, I have another box runs 13272 with the same zoiper without any problem. You can login to our box, I will find you on IRC. On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. ___ 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 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 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_php needed
Anthm, I actually compiled ESL on PHP but wasnt able to figure out how to use it, too little documentation. Can any one throw more light on ESL? Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 15-Jun-09, at 19:29, Anthony Minessale anthony.miness...@gmail.com wrote: Did you actually use ESL with the php wrapper when you tried? You can do all those things from outbound event socket fairly easily. That mod_php you saw, never worked it was just a stub and it didn't actually ever work when the guy who added it totally disappeared, I removed it from tree. And you can still do event socket over localhost on the same box if you so choose. If you really want a mod_php it's entirely possible but it would probably cost you upwards of 5k in development costs. 2009/6/15 Christian Löschenkohl christian.loeschenk...@xpirio.com hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to- get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php ), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: I couldn’t agree more. We’re working with a group that are developing a massive PHP based music application. They are experts in PHP and MySQL but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate with the FS event socket, allows them to work on the areas they know best and not worry about the telephony side too much. We went the lua route, and don’t use the dial plan at all. My vie w is to keep all db access and processing out of FS as much as possible. With the event socket you simply don’t need to embed anything apart fro m the essentials. We are now processing 100,000+ call setups a day (4 hours) per server all using php scripts to drive the application. We may well ultimately use C++ instead of PHP for the event socket comms, but right now PHP does just fine. Regards --- - *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* 13 June 2009 21:57 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] mod_php needed Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net http://irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com hello i am working for an austrian voip carrier. for a few months i work with
Re: [Freeswitch-users] mod_php needed
i tried to
i think i tried everthing and looked closely to everything in libs/esl/php (of
course i build it and included the ESL.php file)
but i do not get the idea in complete, does i work in a client-server way or
in inbound mode like i want to (that is exactly my point)
no examples are there (i would put them in the wiki if i had one)
some simple code i would expect wot work, but i doesn't
?php
require_once(ESL.php);
$esl = new eslConnection('127.0.0.1', '8021', 'asgag243tsa');
$esl-execute(setVariable, codec_string=PCMA);
$esl-execute(answer);
$esl-execute(sleep, 2);
$esl-execute(streamFile,
/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello);
$esl-execute(hangup, 16);
?
can you please help me, what do i get wrong?
br
On 2009-06-15 15:59, Anthony Minessale wrote:
Did you actually use ESL with the php wrapper when you tried?
You can do all those things from outbound event socket fairly easily.
That mod_php you saw, never worked it was just a stub and it didn't
actually ever work
when the guy who added it totally disappeared, I removed it from tree.
And you can still do event socket over localhost on the same box if you
so choose.
If you really want a mod_php it's entirely possible but it would
probably cost you upwards
of 5k in development costs.
2009/6/15 Christian Löschenkohl christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
hi
thank you very much for your input
i can say for me that i realy tried hard to use the event socket
library,
but untill now i can't use it like i used all the agi scripts or
even mod_perl now.
what i do most - in examples, if the server get's an incomming call
- find the right user for the number (not that easy because of did
in austria),
from database or file
- build the right dial string for the bridge application (here i
miss all the php
string functions most)
- unsing mod_php functions like setVariable, getVariable, answer,
transfer, sleep
(i don't see how to do this with the php esl)
- or i check if the number is part of a conferencing product and
build the right
conference setup
i think this would also be possible with lua and luasql, but i
developed years with
phpagi und i'm very used to php in every kind of scripting or
how-to-get-a-solution
situation (since over 10 years now).
for me in our setup it's also the highest goal to get the servers
mostly independent
of each other. i think nobody of our costumers should be unreachable
because a central
scripting/event server or also database server has gone away (as
developers this happens
more often as we would like it to :-))
do not get me wrong, freeswitch is very powerfull and in the near
future it will replace
nearly all of our asterisk servers.
in combination with php the freeswitch plattform would be heaven for me
i also thought Brian Fertig has some source written (as posted on
http://wiki.freeswitch.org/wiki/Mod_php),
in combination of the mod_python rewrite (page was last modified in
june 2007).
br
On 2009-06-14 01:15, Nik Middleton wrote:
I couldn’t agree more. We’re working with a group that are
developing a
massive PHP based music application. They are experts in PHP and
MySQL
but not in VOIP/Telephony. By tuning an abstraction layer that
uses PHP
to communicate with the FS event socket, allows them to work on the
areas they know best and not worry about the telephony side too much.
We went the lua route, and don’t use the dial plan at all. My
view is
to keep all db access and processing out of FS as much as
possible. With
the event socket you simply don’t need to embed anything apart
from the
essentials.
We are now processing 100,000+ call setups a day (4 hours) per server
all using php scripts to drive the application. We may well
ultimately
use C++ instead of PHP for the event socket comms, but right now PHP
does just fine.
Regards
*From:* freeswitch-users-boun...@lists.freeswitch.org
mailto:freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of
*Michael Collins
*Sent:* 13 June 2009 21:57
*To:* freeswitch-users@lists.freeswitch.org
mailto:freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] mod_php needed
Perhaps you should look at controlling calls via the FreeSWITCH event
socket instead of from the dialplan. The nice thing about the event
socket is that your call
Re: [Freeswitch-users] Zoiper reject freeswitch calls
Brain, You are not on irc right now and it is midnight so I'm gona to sleep. however when I try to reproduce that I found it event didn't get to Zoiper. I use the same Zoiper login to two boxes at the same time, version 13272 is ok while the other isn't. I noticed there is an extra line on the log of version 13524: version 13272: 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/839 SOFIA INIT 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/839) State Change CS_INIT - CS_ROUTING version 13524: 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/637 SOFIA INIT 2009-06-15 22:50:46 [DEBUG] sofia_glue.c:1599 sofia_glue_do_invite() sip:6...@192.168.1.27:5070 %3 Setting proxy route to sofia/internal/637 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/637) State Change CS_INIT - CS_ROUTING And here is a more detailed paste: http://pastebin.freeswitch.org/9386 Thank you taking time for this. If you need more detail I'd like to collect and can open ssh for further debug. 7. On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. ___ 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 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 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_php needed
look at the perl examples they should translate to php as all the objects
and methods are the same.
Does anyone who uses ESL + scripting have any time to toss up some wiki
pages?
2009/6/15 Christian Löschenkohl christian.loeschenk...@xpirio.com
i tried to
i think i tried everthing and looked closely to everything in libs/esl/php
(of
course i build it and included the ESL.php file)
but i do not get the idea in complete, does i work in a client-server way
or
in inbound mode like i want to (that is exactly my point)
no examples are there (i would put them in the wiki if i had one)
some simple code i would expect wot work, but i doesn't
?php
require_once(ESL.php);
$esl = new eslConnection('127.0.0.1', '8021', 'asgag243tsa');
$esl-execute(setVariable, codec_string=PCMA);
$esl-execute(answer);
$esl-execute(sleep, 2);
$esl-execute(streamFile,
/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello);
$esl-execute(hangup, 16);
?
can you please help me, what do i get wrong?
br
On 2009-06-15 15:59, Anthony Minessale wrote:
Did you actually use ESL with the php wrapper when you tried?
You can do all those things from outbound event socket fairly easily.
That mod_php you saw, never worked it was just a stub and it didn't
actually ever work
when the guy who added it totally disappeared, I removed it from tree.
And you can still do event socket over localhost on the same box if you
so choose.
If you really want a mod_php it's entirely possible but it would
probably cost you upwards
of 5k in development costs.
2009/6/15 Christian Löschenkohl christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
hi
thank you very much for your input
i can say for me that i realy tried hard to use the event socket
library,
but untill now i can't use it like i used all the agi scripts or
even mod_perl now.
what i do most - in examples, if the server get's an incomming call
- find the right user for the number (not that easy because of did
in austria),
from database or file
- build the right dial string for the bridge application (here i
miss all the php
string functions most)
- unsing mod_php functions like setVariable, getVariable, answer,
transfer, sleep
(i don't see how to do this with the php esl)
- or i check if the number is part of a conferencing product and
build the right
conference setup
i think this would also be possible with lua and luasql, but i
developed years with
phpagi und i'm very used to php in every kind of scripting or
how-to-get-a-solution
situation (since over 10 years now).
for me in our setup it's also the highest goal to get the servers
mostly independent
of each other. i think nobody of our costumers should be unreachable
because a central
scripting/event server or also database server has gone away (as
developers this happens
more often as we would like it to :-))
do not get me wrong, freeswitch is very powerfull and in the near
future it will replace
nearly all of our asterisk servers.
in combination with php the freeswitch plattform would be heaven for
me
i also thought Brian Fertig has some source written (as posted on
http://wiki.freeswitch.org/wiki/Mod_php),
in combination of the mod_python rewrite (page was last modified in
june 2007).
br
On 2009-06-14 01:15, Nik Middleton wrote:
I couldn’t agree more. We’re working with a group that are
developing a
massive PHP based music application. They are experts in PHP and
MySQL
but not in VOIP/Telephony. By tuning an abstraction layer that
uses PHP
to communicate with the FS event socket, allows them to work on
the
areas they know best and not worry about the telephony side too
much.
We went the lua route, and don’t use the dial plan at all. My
view is
to keep all db access and processing out of FS as much as
possible. With
the event socket you simply don’t need to embed anything apart
from the
essentials.
We are now processing 100,000+ call setups a day (4 hours) per
server
all using php scripts to drive the application. We may well
ultimately
use C++ instead of PHP for the event socket comms, but right now
PHP
does just fine.
Regards
*From:* freeswitch-users-boun...@lists.freeswitch.org
mailto:freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf
Of
*Michael Collins
Re: [Freeswitch-users] mod_php needed
Any suggestions of what would be a good example in PHP using ESL to document? I'll take a stab at writing something up this week but it would help to have some idea what would be useful. I've used it and got it working but rather document a generic real life example versus my unique use cases. -- W ___ 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_php needed
A good start would probably be: example of making an inbound connection from script to FS and execute a FSAPI command, like status or show channels. example of making an inbound connection and listening for events and printing them serialized. example of an outbound socket connection where the call is answered, a variable is set then perhaps play one of the pre-installed files and hangup. That last one could be demonstrated using a native socket server or by using ivrd, a little mini forking daemon I added to listen for socket outbound calls and determine a script from channel variables and call that script assuming to use stdin/stdout as the socket. (kinda like agi's) I think that if everyone pooled their experienced together you could probably produce a wrapper that would allow you to use some of your legacy agi code with ESL, naturally you would have to change the names of the apps and a few other things but there is a lot to build on here. I left this portiion of the system where it is so that the community and how it's most commonly used will drive the direction the top layer of code takes. On Mon, Jun 15, 2009 at 10:12 AM, William Suffill william.suff...@gmail.com wrote: Any suggestions of what would be a good example in PHP using ESL to document? I'll take a stab at writing something up this week but it would help to have some idea what would be useful. I've used it and got it working but rather document a generic real life example versus my unique use cases. -- W ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Zoiper reject freeswitch calls
Hi, the difference is the Contact where the %3B should be ; . Is it configurable or a bug? 13272: Call-ID:OTg4NmRlNzY5OThmNzgwM2E3ZmRkYzVhNjVmODMyYjA. User: 8...@192.168.1.15 Contact:user sip:8...@192.168.1.27:5070;rinstance=0f6c69c813f97b49;transport=UDP Agent: Zoiper rev.3065 Status: Registered(UDP)(unknown) EXP(2009-06-16 00:54:12) Host: pbx3.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 839 Auth-Realm: 192.168.1.15 13524: Call-ID:NWJhYWY0YjJmMzdlNWQ4MWIwZjc2NGM5NjQzZDU3NTg. User: 6...@192.168.1.16 Contact:user sip:6...@192.168.1.27:5070;rinstance=cb5d34523fca3400;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3Dcb5d34523fca3400%3Btransport%3DUDP Agent: Zoiper rev.3065 Status: Registered(UDP-NAT)(unknown) EXP(2009-06-16 01:42:56) Host: pbx1.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 637 Auth-Realm: 192.168.1.16 On Jun 15, 2009, at 10:21 PM, Brian West wrote: To: user sip:6...@192.168.1.27:5070;rinstance=59e15734a404c038 Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. ___ 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 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 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] Live Upgrade Techniques
On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo m...@giagnocavo.netwrote: Well, Nextone for instance has a database the keeps most of the state of calls, and it’s replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there’s any kind of switchover. BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve ___ 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] Live Upgrade Techniques
Yeah I was missing this word: SCM = Stateful Call Migration. _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Steve Kurzeja Sent: Monday, June 15, 2009 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo m...@giagnocavo.net wrote: Well, Nextone for instance has a database the keeps most of the state of calls, and it's replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there's any kind of switchover. BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve ___ 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] How do I get a 180 ringing to be sent to an inbound call ?
I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin richard.lam...@mettonigroup.com * Please consider the environment before printing this e-mail * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN * ___ 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_php needed
hi could you provide me a simple example? - connect with esl - get uuid - set a variable (e.g. codec_string=PCMA) - answer the channel - playback a file the script ist called from ivrd, if i get it right in the dialplan it's action application=set data=ivr_path=/opt/freeswitch/scripts/test.php/ action application=socket data=127.0.0.1: full/ with ivrd started as ./ivrd -h 127.0.0.1 -p in my setup $esl-api(help) works and also $esl-sendRecv(api help) but $esl-execute() does nothing i use version 1.0.4pre8 if it is helpfull br On 2009-06-15 17:12, William Suffill wrote: Any suggestions of what would be a good example in PHP using ESL to document? I'll take a stab at writing something up this week but it would help to have some idea what would be useful. I've used it and got it working but rather document a generic real life example versus my unique use cases. -- W ___ 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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ 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] How do I get a 180 ringing to be sent to aninbound call ?
Survey says ... execute the ring_ready application /b On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote: Something to consider is how long will be PSTN allow the call to remain un-answered. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Richard Lamkin Sent: Monday, June 15, 2009 11:28 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I’ve looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin richard.lam...@mettonigroup.com * Please consider the environment before printing this e-mail * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN * ___ 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 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] Gateway conference handshake trouble
Hi, We are trying to work freeswitch into an older system as a conference bridge. Our existing gateways can hand off the call and they pass some DTMF signals to route everything. Currently, the gateway sends * and then freeswitch returns a # to accept the call for the conference. When that transaction is done the gateway waits 2 seconds to send either a 0 or 1 and indicate if the caller is a moderator/admin of the conference. Everything works until we get to the moderator flag. That part never appears in the logs or debug info in the console. But the person on the call, can hit 0 or 1 and then enter the conference call correctly. If they don't it will timeout and drop them. From what we know the gateway is sending the moderator DTMF flag, but freeswitch is only listening on the call for it. Any idea would appreciated. Lon ___ 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] Access MySQL directly via Javascript using SSJS Engines .... really!
You know you can already access any sql support by UnixODBC via mod_spidermonkey already... And NO its not slow (maybe that was true 10 years ago, but not any longer) From: Stephen Crosby stevecr...@gmail.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 15 Jun 2009 14:51:30 -0700 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines really! What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl edpime...@gmail.com wrote: FYI Now you can directly connect to MySql via javascript http://jsext.webloji.net/ http://jsext.sourceforge.net/JSEXT1.Mysql.html http://www.brainonfire.net/blog/ssjs-on-ubuntu/ And if you wish you had a V8, check out Google SSJS Engine BTW: JS is now 7x faster than it last year. -E http://Gpro.ws http://twitter.com/edpimentl http://facebook.com/facevalu ___ 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 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 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] Access MySQL directly via Javascript using SSJS Engines .... really!
I'm actually much more interested in the HTTP library and a few other components than MySQL. Freeswitch's spidermonkey CURL library doesn't provide returned HTTP status codes and JSEXT does. That said, I'm still somewhat interested in the mysql library over odbc. For me, the only thing I've ever really done with ODBC is use it to bridge the gap between freeswitch mod_spidermonkey and mysql server. I think it would be nice to not need it. --Stephen On Mon, Jun 15, 2009 at 2:58 PM, Ken Rice kr...@freeswitch.org wrote: You know you can already access any sql support by UnixODBC via mod_spidermonkey already... And NO its not slow (maybe that was true 10 years ago, but not any longer) -- *From: *Stephen Crosby stevecr...@gmail.com *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Mon, 15 Jun 2009 14:51:30 -0700 *To: *freeswitch-users@lists.freeswitch.org *Subject: *Re: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines really! What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl edpime...@gmail.com wrote: FYI Now you can directly connect to MySql via javascript http://jsext.webloji.net/ http://jsext.sourceforge.net/JSEXT1.Mysql.html http://www.brainonfire.net/blog/ssjs-on-ubuntu/ And if you wish you had a V8, check out Google SSJS Engine BTW: JS is now 7x faster than it last year. -E http://Gpro.ws http://twitter.com/edpimentl http://facebook.com/facevalu ___ 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 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 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 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] Access MySQL directly via Javascript using SSJS Engines .... really!
On Jun 15, 2009, at 5:08 PM, Stephen Crosby wrote: I'm actually much more interested in the HTTP library and a few other components than MySQL. Freeswitch's spidermonkey CURL library doesn't provide returned HTTP status codes and JSEXT does.\\\ Patch it! ;) That said, I'm still somewhat interested in the mysql library over odbc. For me, the only thing I've ever really done with ODBC is use it to bridge the gap between freeswitch mod_spidermonkey and mysql server. I think it would be nice to not need it. --Stephen ___ 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] Access MySQL directly via Javascript using SSJS Engines .... really!
Actually these new SSJS engines such GoogleV8 and other such as JAXER Bring a entire new way of building robust webapp/desktop app/ mobile app like it has never been built before... For those that love Google GWT=Java_To_Javascript and dislike verbosity of Java, there is PyJamas ... Google GWT=Python_To_Javascript JS today is not the same it was years ago or even months ago. -E ___ 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] Access MySQL directly via Javascript using SSJS Engines .... really!
I don't think V8 will work on 64bit yet will it? /b On Jun 15, 2009, at 5:45 PM, EdPimentl wrote: Actually these new SSJS engines such GoogleV8 and other such as JAXER Bring a entire new way of building robust webapp/desktop app/ mobile app like it has never been built before... For those that love Google GWT=Java_To_Javascript and dislike verbosity of Java, there is PyJamas ... Google GWT=Python_To_Javascript JS today is not the same it was years ago or even months ago. -E ___ 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] Force SIP UA to pick up call during ringing?
I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter ___ 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] is there any available gui for freeswitch using cake php?
I like to use this GUI for both OS windows and linux. Wikipbx and PFSENSE is for linux only... Just a simple website that I need to integrate... seven-8 wrote: On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? AFAIK, no fully developed GUI available yet, just curious, why are you finding a GUI instead of wikipbx or pfsense? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24045733.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] How can I join two freeswitch on two servers?
I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Error on mod_nibblebill cannot connect to ODBC
Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER= root PASS= password . I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Is Freeswitch ready for prime time?
Hehe, where can I buy stock in this company? :) -MC Sent from my iPhone On Jun 15, 2009, at 4:33 AM, David Knell d...@3c.co.uk wrote: On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? Hi Paul - We've used FS over the last 18 months or so to handle millions of calls - some wholesale in/out, some IVR, some calling card, some callthrough - with a total value in the millions of dollars; we have no complaints. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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 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] is there any available gui for freeswitch using cake php?
I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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] Access MySQL directly via Javascript using SSJS Engines .... really!
As of April 09 it did not support 64bit not sure if it has been added since then. -E ___ 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] is there any available gui for freeswitch using cake php?
Can you share me the link of it so i can try... Please Diego Viola wrote: I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz darklio...@yahoo.com wrote: Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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 -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Access MySQL directly via Javascript using SSJS Engines .... really!
Pretty useless without 64bit support. /b On Jun 15, 2009, at 9:58 PM, EdPimentl wrote: As of April 09 it did not support 64bit not sure if it has been added since then. -E ___ 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
