Re: [Freeswitch-users] Performance Tuning
libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles If you only have one provider for your trunk is it possible to set up multiple profiles for enhanced performance? For example if I have multiple DDIs from the provider can I set up a different profile for each one? Or maybe based on some some sort of a pattern? ___ 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] Question regarding running FreeSWITCH with high priority enabled.
On Tue, Dec 8, 2009 at 5:42 AM, DJB djbin...@yahoo.com wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. If you could explain how you are doing the load balancing it would be really helpful to me. I am trying to do the same thing. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Rocks!!!!!!!!!
Preface 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. 2. Choosing hardware options (server, phones, gateways) 3. Setting up FS 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS gateways, etc.) 5. Administering FS (CLI and GUI) 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) 7. Performance, sound quality, other issues 8. Writing scripts (LUA, etc.), connecting to databases 9. Real-life examples (Gino's Pizza, etc.) Conclusion Index -- I found the rosetta stone useful though woefully lacking in volume. I guess that's true overall with the project. ___ 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] Even socket question.
Hey All. I am trying to get freeswitch to route to my socket handler and am having a problem. I am running freeswitch inside a virtualbox VM for testing purposes. The vitualbox communicates with my host via the host only adapter. The VM IP address is 192.168.56.3 and the laptop has the iP 192.168.56.1 I have set up both an outbound and an inbound socket handlers. The inbound one works fine, the outbound is not working . The inbound merely logs the event name. The outbound logs the connection and hangs up. I have set up an extension like this include extension name=8084 condition field=destination_number expression=^8084$ action application=set data=continue_on_fail=true / !-- we still need this to continue if bridging times out -- action application=set data=call_timeout=5 / action application=socket data=192.168.56.1:8084 sync full/ /condition /extension /include When I dial 8084 I get a lot of events being logged but the oubound never gets the calls and never logs the call. I have added the fs_cli output below. It looks to me like it's sending the output to the other IP address of my laptop instead of the one I specified in my extension but I could just be misreading that. I have set the external IP of the freeswitch to the 56.3 address. Here is the LSOF output freeswitc 2468 root 31u IPv4 5785 TCP ubuntuvm01:5080 (LISTEN) freeswitc 2468 root 33u IPv6 5791 TCP localhost:5060 (LISTEN) freeswitc 2468 root 36u IPv4 5804 TCP 192.168.56.3:5060 (LISTEN) freeswitc 2468 root 48u IPv4 5910 TCP 192.168.56.3:8021 (LISTEN) freeswitc 2468 root 50u IPv4 5912 TCP *:8080 (LISTEN) Here is the output from the fs_cli 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl domains. Falling back to Digest auth. 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl domains. Falling back to Digest auth. 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1...@192.168.56.3 [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa] 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1...@192.168.56.3) Running State Change CS_NEW 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1...@192.168.56.3) State NEW 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel sofia/internal/1...@192.168.56.3 entering state [received][100] 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP: v=0 o=Z 0 0 IN IP4 218.101.6.157 s=Z c=IN IP4 218.101.6.157 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:115:32000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:107:16000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G722:9:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMU:0:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMA:8:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec sofia/internal/1...@192.168.56.3 GSM/8000 20 ms 160 samples 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload to 101 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885 (sofia/internal/1...@192.168.56.3) State Change CS_NEW - CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1...@192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1...@192.168.56.3) Running State Change CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1...@192.168.56.3) State INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83 sofia/internal/1...@192.168.56.3 SOFIA INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111 (sofia/internal/1...@192.168.56.3) State Change CS_INIT - CS_ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1...@192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1...@192.168.56.3) State INIT going to sleep 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314
Re: [Freeswitch-users] Even socket question.
On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale anthony.miness...@gmail.com wrote: do you have something listening on 8084 ? Yes. I figured out the problem. There was already an extension called 8084 and it overwrote the extension I defined. Which brings me back to a question I had earlier. Where is the equivalent of the show dialplan command? How can I list all the extensions and their definitions? ___ 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] Lua and database access to core_db
On Wed, Dec 9, 2009 at 7:56 AM, Kendall Stauffer k...@ksac.com wrote: Hey you guys, I know this isn’t the right place for this, but I have been working with asterisk for 5 years now, and just got freeswitch working (on windows, not os x yet). All I can say is AWESOME --- thanks so much Out of curiosity. Did you choose freeswitch because it runs on windows and asterisk doesn't? I find some people choose freeswitch because they don't know or want to use linux (obviously this doesn't apply to you) and some people choose it because they want a windows solution and asterisk doesn't run on windows. ___ 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] HA questions.
On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote: The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? From an as of now unkown SIP trunk provider (we are still in negotiations with a couple of companies). ___ 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] HA questions.
On Fri, Dec 4, 2009 at 5:56 AM, Adam Ford li...@redbonez.net wrote: Have you checked out Redfone? While I haven't attempted to implement it yet, my Redfone foneBridge2 claims to be able to handle load balancing and failover between two Asterisk/Freeswitch servers. That would be my choice for incoming E1 lines. Right now I am looking for a SIP solution. ___ 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] IAX? Issues connecting road warriors with SIP?
Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) ports fail being opened dynamically to work properly, or does SIP today really work well over NAT firewalls? Yes I get issues quite a bit with the server being behind a firewall. IAX is much nicer in this circumstance. ___ 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] HA questions.
I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to lose those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. ___ 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] XML config file parsing
On Tue, Nov 24, 2009 at 5:48 AM, Eliot Gable egable+freeswi...@gmail.com wrote: Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. That sounds like a pretty sane way to go bout it. ___ 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] XML config file parsing
On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman rob4manh...@gmail.com wrote: Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of time? ___ 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] Hardware echo cancellation.
I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? 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] Hardware echo cancellation.
On Thu, Nov 19, 2009 at 10:04 AM, David Knell d...@3c.co.uk wrote: Hi Tim, Here you go: http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html Thanks. That's almost exactly the same situation as the one I am going to find myself in. In (very) brief: maybe, no, and depends on the definition of 'lots'. By lots I mean somewhere between 50 to a 100 but it's mostly an IVR application so all it will be doing is either playing prompts or recording messages. Almost no live conversations. ___ 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] 1.05
Where is 1.05? The trunk? Is trunk stable? 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] Dialogic cards
On Tue, Jul 14, 2009 at 3:39 AM, Steve Underwoodste...@coppice.org wrote: Tim Uckun wrote: We have some older dialogic cards (D300 series E1 cards) and I am wondering if freeswitch can support these cards. Oh, I like the easy questions. No. It lacks the hardware features to do anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or anything else that expects to do two way telephony through the host CPU. What are the recommended cards to be used with freeswitch? ___ 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] Dialogic cards
We have some older dialogic cards (D300 series E1 cards) and I am wondering if freeswitch can support these cards. 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
