February 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH
by Marcin Gozdalik 17 Feb '13

17 Feb '13

2013/2/11 Eliot Gable <egable+freeswitch(a)gmail.com>: > On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik <gozdal(a)gmail.com> wrote: >> >> +1 >> >> I do not doubt mod_ha is necessary inside of FS and it may be >> better/simpler than writing Pacemaker resource agent, but writing >> yet-another-cluster-communication-engine is IMHO the wrong way to go >> and using Corosync for communication will give a lot of value from >> mature codebase. >> > > I understand what you are saying, but what I am trying to get across is that > I am not writing yet-another-cluster-communication-engine. All I am really > doing is combining a multicast messaging API written by Tony and the event > API in FS to broadcast existing state information between multiple FS nodes, > as well as adding a tiny amount of logic on top of that to coordinate call > fail over and recovery. That's probably a little over-simplified, but it > gets the point across. The network communication code is already in FS and > well tested. The event system is already in FS and well tested. I also think I understand what you are saying. It means we have trouble putting thought into writing ;) >From what I understand what you are trying to achieve is that every node in FS "cluster" knows what are the nodes and whether they are down or up. What I am saying is that this simple task is fundamentally hard. Sending and receiving multicast is easy, but keeping distributed state consistent between nodes in cluster is hard (like in really hard, harder than writing VoIP softswitch all over again), especially in case of Byzantine failures (i.e. nodes lying that they are down when they are up or other way round). I am no big expert in the area but seen at least 2 cases (MMM - http://www.xaprb.com/blog/2011/05/04/whats-wrong-with-mmm/ and Chubby in Google - http://www.read.seas.harvard.edu/~kohler/class/08w-dsi/chandra07paxos.pdf) where people were trying to write (MMM) or use (Chubby) some kind of distributed code and failed. That's why whenever I see anything related to distributed state I say that it's way beyond my understanding and best is to use something that works. I have > already written the code to the point that it parses the configuration files > and starts sending heartbeats out all of the interfaces configured. I have > also already written a lot of the code that deals with the state > transitions. All I am talking about doing is implementing a tiny little > finite state machine. It's a pretty trivial programming task. In fact, I > think it was covered in my first year at Carnegie Mellon University. Of > course, I had already figured out how to write such things in high school, I > just did not know what it was called at that point. My point is, that this > is not yet-another-cluster-communication-engine. It is a very specific and > small finite state machine designed solely with the goal in mind of making > FS have just enough information to coordinate call fail over internally. If > I recall correctly, a lot of people also said writing > yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and > it was obviously worth the effort. And I am not even trying to do something > as complex as that. If you think this is > yet-another-cluster-communication-engine, you are missing the point. It is > not. It never will be. See above - if it will never be and you are trying to achieve distributed, consistent state between nodes, IMHO you are going to get it wrong. Frankly I lack the knowledge and time to check if Corosync API is perfect for this task. As Anthony suggested elsewhere maybe it is possible to abstract the communication/keeping distributed state part so that it would be easy to provide Corosync or other (OpenReplica? Zookeeper?) implementations. > Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name > carrier-grade telecom providers have a built-in solution for automatic call > fail over. The only way FreeSWITCH will ever compete with such solutions is > if it also has that feature. Pacemaker and Corosync are overkill just to get > FS to handle single node failures and provide call recovery. It took me a > full 3 months of working with them every day to really understand how to > deploy them properly in conjunction with FreeSWITCH and Postgres to provide > a carrier-grade hot-standby solution which was robust enough to handle 99% > of the failures I could throw at it. Granted, this was back when the > configuration still needed to be written by hand in XML and prior the > existence of any resource agent for FreeSWITCH. But, even with those > changes, deploying Pacemaker and Corosync is not a simple task. If that is > the requirement for FS to have HA, it will never truly stand a chance > against commercial offerings. I believe that Clusters from Scratch http://clusterlabs.org/doc/en-US/Pacemaker/1.1/html/Clusters_from_Scratch/) allowed me to setup a working Pacemaker/Corosync installation on Debian in less than a day. It is a fair point that FS to compete with big names (at least in marketing buzz-feature checklist) has to have HA. Trouble is HA is always hard. Making it simple and work is the ultimate goal but I'd rather shoot at "work" first and later for "simple". Comparing FS to commercial offerings is hard as the way FS is deployed is usually different than commercial competitors. If you buy Broadsoft you don't get a Debian package or sources to compile - you get a bunch of highly-paid consultants that install and configure everything for you. If HA in FS would work that way it would maybe be even better for building business around FS ;) Jokes aside - if somebody wants to configure FS in HA evironment I don't think it is much of an obstacle for him to configure Corosync as already he has to have some kind of DB failover as well, redundant switches, power, etc. If some commercial vendor is able to manufacture an out-of-the box solutions based on FS with a "HA" checkbox somewhere in Web configuration - well, congratulations to him and he I hope he would be deservedly reaping profits from his product. -- Marcin Gozdalik

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Problems setting up mod_gsmopen
by J G 17 Feb '13

17 Feb '13

Hi! I am trying to figure out how to get my E173 to work with Freeswitch. I accually had it working a few months ago and don't know if I have changed anything to break it. But atleast we know it is proven to have worked and has voice unlocked. The interface registers in gsmopen very randomly, something I have read is a problem for others too. Usually it connects to the provider via USB2, and sometimes USB0, but it is very random. This is not my main problem though, it is to get it working at all. On the (random) ring interface I get this output via picocom if I call the number: RING +CLIP: "+61468440XXX",145,,,,0 That should be the "controldevice_name", right? The other two are silent, but I did get some scrambled output on one, ...once. If I get it to connect to the network, and try to call out, I get nothing on one interface, and an error if I select the other: 19.218876 [ERR] switch_core_io.c:806 gsmopen/ANY/148 has no write codec. What does that error try to explain to me? Sorry for a very messy post, but everything feels so random and I can't even reproduce for good logfiles. Make a call, sighup the logfile, and the next call 10 seconds later the output is something totaly different, it's that random! I can work on proper logfiles if it's needed. - J

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Auto attendant :: act on no digit entry
by Sayyed Mohammad Emami Razavi 17 Feb '13

17 Feb '13

I want to transfer customers to operator(ext=100) on no digit entry after autoattendant with xml IVR! can anybody help me?! Blue.box apparently can not do this task!

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About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT
by sparklezou 17 Feb '13

17 Feb '13

Hi All, The Sangoma A101 board support "hardware fax detection". It configured in /etc/wanpipe/wanpipe1.conf configureation file. TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware Is there any configuration on the dial plan, if detect the fax, goes to the predefined dial plan context. something the same in asterisk http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect Thanks! 2013-02-17 sparklezou

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Re: [Freeswitch-users] FreeSWITCH HA BoF Meetings
by Eliot Gable 16 Feb '13

16 Feb '13

I am usually just getting my kids in bed at 8pm, so 8:15pm is typically better timing. On Feb 15, 2013 7:26 PM, "Michael Collins" <msc(a)freeswitch.org> wrote: > will do. > > On Fri, Feb 15, 2013 at 1:01 PM, Giovanni Maruzzelli <gmaruzz(a)gmail.com>wrote: > >> ouch! at 08:00pm EST will be 02:00am my time. >> Not sure I'll be able to attend the live discussion. Please record it as >> always, please :) >> >> -giovanni >> >> >> On Fri, Feb 15, 2013 at 9:59 PM, Giovanni Maruzzelli <gmaruzz(a)gmail.com>wrote: >> >>> I'm very much interested, thanks Eliot for your efforts so far. >>> >>> -giovanni >>> >>> On Thu, Feb 14, 2013 at 11:41 PM, Michael Collins <msc(a)freeswitch.org>wrote: >>> >>>> Hello! >>>> >>>> Yesterday we had a nice discussion >>>> <http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13>about all things >>>> HA. Eliot Gable gave us a lot of very good information about the state of >>>> mod_ha_development and all the work that he's done over the past 4+ years >>>> in the world of HA and FreeSWITCH. >>>> >>>> It's now time to let those who have a vested interest in HA to get >>>> together with Eliot and continuing the discussion. To that end we propose a >>>> FreeSWITCH HA BoF (Birds of a Feather) conference. Eliot says that he can >>>> be available after 8PM EST. I would like to propose that the meetup be >>>> Tuesday evenings at 8:00PM EST in the main FreeSWITCH public conference. >>>> >>>> If you would like to meetup then please email me off-list and CC Eliot ( >>>> egable+freeswitch(a)gmail.com) so that we can keep track of who is most >>>> interested in this topic. >>>> >>>> Thanks! >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > >

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codec negotiation weirdness
by Brian J. Tarricone 16 Feb '13

16 Feb '13

Hi all, I'm experiencing some strange behavior with codec negotiation. I'm using the default early negotiation, and I'm setting inbound-codec-negotiation to "scrooge" so FS will always have the final word on what codec is used. (I've also tried "greedy" with the same results. "generous" does not meet my needs, unfortunately.) I'm setting inbound-codec-prefs to the following: opus@48000h@20i,isac@32000h@30i,speex@16000h@20i,speex@8000h@20i,PCMU However, I'm seeing FS select PCMU for a client that advertises both isac/32k and PCMU. The relevant portion of the FS log is below. 2013-02-16 01:45:06.947647 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 1772524861 1772524861 IN IP4 10.70.209.126 s=- c=IN IP4 10.70.209.126 t=0 0 m=audio 57178 RTP/AVP 103 104 0 8 106 105 13 126 c=IN IP4 10.70.209.126 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[opus:116:48000:20:32000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[SPEEX:99:16000:20:42200] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:106:32000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:105:16000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [CN:13:8000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [telephone-event:126:8000:20:0]/[SPEEX:99:8000:20:24600] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5268 Set 2833 dtmf send/recv payload to 126 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:104:32000:30:0]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-16 01:45:06.947647 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! Sticking with PCMU@8000h@20i As you can see, it first compares all offered codecs against opus/48k, next skips isac completely, then goes to speex/16k, speex/8k, and PCMU, and eventually grudgingly accepts PCMU. Interestingly, if I drop opus from the list, and leave the rest as-is, it does select isac/32k, but weirdly: 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5139 Audio Codec Compare [ISAC:103:16000:30:0]/[isac:99:32000:30:160000] 2013-02-16 01:51:10.691314 [DEBUG] sofia_glue.c:5161 Bah HUMBUG! Sticking with isac@32000h@30i It picks isac/32k after only examining the isac/16k line in the SDP, before it even gets to isac/32k. (In fact those are the only 2 lines after the SDP is printed; it makes the decision after making just one comparison.) Any ideas? Am I just misunderstanding how this works? Thanks, Brian

1 1

WebRTC to SIP
by Jack 16 Feb '13

16 Feb '13

Has anyone been successful in connecting to freeswitch with this? http://kapejod.org/webrtc/sip-audio-demo.html I am able to register on freeswitch and place a call to the webpage, which rings, but no audio. This is for Canary Beta browser only right now. Jack

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Prolem with SIP OPTION ping
by Gary Chen 16 Feb '13

16 Feb '13

Version: Freeswitch 1.2.5.3 with ODBC as core db. For redundancy purpose, I have two FS running behind a Kamailio SIP proxy. My Kamailio uses diapatcher to send call to FS1 first. If FS1 is not available, Kamailio will send call to FS2. Kamailio frequently send out SIP OPTION message to check each FS. If no reponse back from one of FS, it will mark it as Inactive and stop sending call to it. The problem I have is that when the odbc connection gone bad, the FS is still up and running and it still reply back "200 OK" to SIP OPTION from Kamailio. So Kamailio dispatcher will not mark it as Inactive and continue sending call to the bad FS and call will fail. Is there anyway I can make FS not reponse back to Kamailio with "200 OK" when core ODBC is down? Or shutdown the FS when core ODBC is down? Gary

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http://infiltrated.net
by Brian West 16 Feb '13

16 Feb '13

Does anyone know if these lists are updated still? -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

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About timer X fired?
by yyu0378 15 Feb '13

15 Feb '13

hi: My fs send call from one gatway to another,but in cli has many like timer X fired.If this is my seesion-timeout is wrong.The seesion-timeout defaults is 30 minutes.Other gateways is one hour.Did i change it?Thank you !!

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