September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN
by James Mortensen 13 Sep '13

13 Sep '13

Hello, I'm working on a one-way audio bug that may or may not be either Asterisk or Chrome WebRTC related. Details are here: https://code.google.com/p/webrtc/issues/detail?id=2347, but basically the problem is that I get about 1 in 10 calls returning one way audio where audio flows from the PSTN to Chrome but not the other way. I believe the issue may be with Asterisk 11, based on Wireshark traces and other troubleshooting, and if that's the case, I'm happy to try Freeswitch as an alternative. I tried the demo here: https://webrtc.freeswitch.org/webrtc/portal.html, and the 20 test calls to the PSTN worked perfect with two way audio. So, now, onto the Freeswitch question: Where is all the documentation for configuring the Websocket server on Freeswitch 1.4b? Where is the documentation that explains how to get ICE enabled? Why, when I enabled the ws-bind port in internal.xml does the client fail to establish a WS connection? Basically, I'd love to get my hands on the docs for setting this up, but the Internet is largely silent on this matter? Is it possible for the Freeswitch team to share the configuration they used to get that JsSIP/Freeswitch demo setup? If not the configuration, where would I find the docs for Freeswitch 1.4b? Thank you!! James *"Every Call, Every Time!"* How did I do? [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735> Synclio Extension:* 110627* Phone: * 866-707-4590* Skype: *No!* *Call me on Synclio! :)* - "*Another great thing about keeping the bug count at zero is that you can respond much faster to competition. Some programmers think of this as keeping the product ready to ship at all times." - Joel Spolsky, Author of Smart and Gets Things Done*

4 15

Friday Call… don't forget to join in!
by Brian West 13 Sep '13

13 Sep '13

Don't forget the Friday Free For all is on... I'm here all alone waiting for you... how ever creepy that sounds! :P LOL -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

3 2

Delay
by Blake Priddy 13 Sep '13

13 Sep '13

Fixed (drops microphone, walks off stage) :) Don't use Epygi as a gateway for your PRI Adtran FTW

1 0

Eavesdrop results in rewriting RTP sequence numbers on a bridge.
by Richard Brady 13 Sep '13

13 Sep '13

Hi guys Seems than running eavesdrop (and potentially other media bugs) on a bridged channel results in the rewriting of sequence numbers on RTP streams, potentially hiding packet loss when it occurs and creating havoc with downstream jitterbuffers. Is there a way to disable this behaviour? Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

4 9

Chat issue
by Vilson C. Gärtner 13 Sep '13

13 Sep '13

Hi people, I'm subscribed to this list a few weeks and this is my first post. So, let me first thank the FS folk for this great project. I've a little background in Asterisk, but FS surprised me in many points. So, as I told I'm new on FS and have been playing around for few days. Even so, thanks to the well documented help and a lot of Google searching, I was able to make things working like a charm: video, inbound, outbound... everything is working nicely. The problem I am struggling against now is send chat messages from one extension to another. I'm pretty sure I'm missing something (obvious), but I can't solve it. Here is what happens: my sever as an IP (lets suppose 123.123.123.123) , but the extensions are registered using the server's name (voip2.xxxxx.xxx). Using the cli command, I've found out the problem: sending a "chat sip" to an extension, using the IP, works without problem, for instance: chat sip|1001@123.123.123.123|1007@123.123.123.123|Message But when I use the server's name, I see the following errors: [WARNING] sofia_presence.c:219 Not sending to local box for 1007(a)voip2.xxxxx.xxx [ERR] sofia_presence.c:270 Chat proto [global] from ["1001" <sip:1001@voip2.xxxxx.xxx>] to [1007(a)voip2.xxxxx.xxx] Mensagem enviada (Message sent) !!!! (not true ;-) Nobody to send to: Profile voip2.xxxxx.xxx Sending a message from one extension to another, the error says: Nobody to send to: Profile internal I added an alias to internal.xml, without results. To finish my first (and huge) message, let me say that when I register an extension using the server's IP it works. So, what must I configure and where, to solve this? Thank you. Best regards, Vilson

1 0

Re: [Freeswitch-users] mod_managed version
by Ed Graves 13 Sep '13

13 Sep '13

Thanks Dave, Thanks Peter. Right after getting Peter's reply, I told management here that we should upgrade to VS2012, so that's what I'm using now. I'm to the point where FreeSWITCH is looking for my mod_managed script, but I don't know how to create one. The documenation says, "mod_managed should "just work" with the .NET Framework 3.5 and Visual Studio 2008 installed. In the build configuration manager inside Visual Studio, mod_managed has configurations for both Debug/Release CLR and Mono. Simply select Debug_CLR and build." I don't see any "build configuration manager". Is this different in VS2012? I cannot figure out how to load the FreeSWITCH libraries in my application. Again, any help would be appreciated.

1 0

Re: [Freeswitch-users] FreeSWITCH freezes when upgrading to 1.2 branch - SOLVED
by Lloyd Aloysius 13 Sep '13

13 Sep '13

Tony, Thank you for the fix. This will help a lot. Ashley and Regis, I would like to share some information about this bug. I reported in july http://jira.freeswitch.org/browse/FS-5582 and Tony find the reason why this is happening. Then I went to ClueCon. This is my first time in ClueCon. Before the conference night in the Pizza party, I met Tony and introduce myself. He then introduce the Zoper developer and explain this problem. --- I went with so many questions in my mind before ClueCon. I got all the answers by end of ClueCon. Tony and Team - Thank you very much for such a Great Conference and help all the time. Thank you Lloyd On Fri, Sep 13, 2013 at 2:46 AM, Regis M <regis.freeswitch.org(a)tornad.net>wrote: > Hi, > Thanks a lot for this bug reporting. > We need more bug/problems solved report like this. It is very interesting > and important to find help when we try to solve similar bugs and find > people who report a clear and factual solution to a problem and not only > bug or asking help with big thread with a lot a suggestion... > > Thanks a lot and a nice work to solve this..!! :). > > Regards, > > > > > > > 2013/9/13 Anthony Minessale <anthony.minessale(a)gmail.com> > >> yes >> >> >> On Thu, Sep 12, 2013 at 9:50 PM, Ashley Breeden <ash(a)url.net.au> wrote: >> >>> Hi Anthony, >>> >>> Thanks for doing that so quickly, greatly appreciated. >>> >>> Is it possible to have this patch added to the next 1.2 stable release? >>> >>> Ash. >>> >>> On 13/09/2013, at 1:23 AM, Anthony Minessale < >>> anthony.minessale(a)gmail.com> wrote: >>> >>> I checked the code and found the problem. Zoiper was not happy to get a >>> NOTIFY with Expires: 0 after sending a SUBSCRIBE with Expires: 0 >>> I added a patch to the code to skip responding with such a NOTIFY when >>> zoiper is present. >>> >>> This is not a real fix but its better than having everyone's box die. >>> >>> >>> >>> On Thu, Sep 12, 2013 at 9:40 AM, Ashley Breeden <ash(a)url.net.au> wrote: >>> >>>> Hi All, >>>> >>>> I wanted to share my findings with everyone on the list in case anyone >>>> runs into the same issue I had as it what a nightmare to find. A special >>>> thanks must be given to FreeSWITCH Consulting, especially Eliot who >>>> assisted in debugging and giving me some fresh ideas to look at. >>>> >>>> For some time now I have struggled to upgrade my servers to 1.2 stable >>>> from 1.0.head (git-e3698c2 2012-03-23 01-21-55 +0100). The issue I always >>>> faced was is that after I upgraded the servers it would run happily for the >>>> next 5-10 hours but for no reason FreeSWITCH would become unresponsive and >>>> stop accepting registrations. If you ran command like show channels, >>>> status etc the command would just freeze and there would be no response. >>>> >>>> Typically the freeze would occur when the servers were practically >>>> idle, looking at the logs you would see: >>>> >>>> [WARNING] sofia_reg.c:1524 SIP auth challenge (REGISTER) on sofia >>>> profile 'internal' for [1050(a)domain.com] from ip x.x.x.x >>>> [WARNING] sofia.c:1639 MSG Thread 2 Started >>>> [WARNING] sofia.c:1639 MSG Thread 1 Started >>>> >>>> In attempt to resolve it I tried new hardware, changed from Centos to >>>> Debian, MySQL to Postgres, nothing I seemed to do fixed the issue. >>>> >>>> I started to run some network analysis on the server and what I noticed >>>> is about 20 minutes before the server freeze the traffic would spike from >>>> 100kbps to around 400kbps, on closer inspection I noticed that the 300kbps >>>> spike was actually between the server and a single IP on port 5060. >>>> Analysing the packets (tcpdump -s0 -A port 5060) I could see that one >>>> registration was sending SUBSCRIBE events at the rate of about 300-400 per >>>> second and FreeSWITCH was sending a NOTIFY back for each SUBSCRIBE, once >>>> the server froze the traffic would stop which is what made it hard to find >>>> as you had to be looking at the network traffic before the freeze. Once I >>>> knew the user causing the increased traffic flow I was able to work out >>>> that the user agent causing this was Zoiper (rev. 11137). It would appear >>>> Zoiper would get into a loop with the server and keep sending SUBSCRIBE >>>> packets for the message waiting indicator. >>>> >>>> I have spoken to Zoiper who have taken the issue on board and informed >>>> me it does not happen on Zoiper 3, but does happen on Zoiper 2, however >>>> they plan to fix it in the next release of Zoiper 2. They suggested for >>>> the moment to disable the MWI Subscribe option by doing "Go to the >>>> preferences -> select your sip account -> enable "show advanced options" >>>> andchange the mwi subscribe drop down to "disabled" >>>> >>>> The strange thing with all this is that the same Zoiper client never >>>> had a problem on version 1.0.head (git-e3698c2 2012-03-23 01-21-55 +0100) >>>> but as soon as you upgraded to 1.2 or even the current git versions you >>>> would find that FreeSWITCH and Zoiper would get into a SUBSCRIBE/NOTIFY >>>> Loop. >>>> >>>> I hope this information is of some help to someone else in the future >>>> and is able to ensure they do not have to go through the same nightmare I >>>> did. >>>> >>>> If anyone has any suggestions on ways to limit the SUBSCRIBE/NOTIFY >>>> responses so that you can avoid this happening in the future I would be >>>> very interested to hear, even if you have any other feedback that might >>>> have assisted in finding this issue quicker I would be very interested. >>>> >>>> >>>> Regards, >>>> >>>> *Ashley Breeden* >>>> URL Networks Pty Ltd >>>> *WWW:* http://www.url.net.au >>>> *Twitter: *@urlnetworks >>>> >>>> * >>>> * >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale@hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888@conference.freeswitch.org >>> googletalk:conf+888@conference.freeswitch.org >>> pstn:+19193869900 >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> 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 >> >> > > _________________________________________________________________________ > 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 > >

1 0

Unreliable DTMF
by Alex Lake 13 Sep '13

13 Sep '13

I've got an application that doesn't seem to work well with DTMF coming in (supposedly using rfc2833) from some phones - notably Samsung S3s. I have told my carrier to use rfc2833. I have put dtmf-type="rfc2833" into my SIP profile. I see "a=fmtp:101 0-16" in the session progress message I was wondering what this message "Activate Buggy RFC2833 Mode" means! When a normal user hits a DTMF key, we get this in the log: 2013-07-25 13:26:19.308267 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:744 2013-07-25 13:26:19.308267 [DEBUG] switch_channel.c:471 RECV DTMF 1:744 Does this all indicate that rfc2833 is working?! Anyone else had trouble with S3s?

6 12

FreeSWITCH freezes when upgrading to 1.2 branch - SOLVED
by Ashley Breeden 13 Sep '13

13 Sep '13

Hi All, I wanted to share my findings with everyone on the list in case anyone runs into the same issue I had as it what a nightmare to find. A special thanks must be given to FreeSWITCH Consulting, especially Eliot who assisted in debugging and giving me some fresh ideas to look at. For some time now I have struggled to upgrade my servers to 1.2 stable from 1.0.head (git-e3698c2 2012-03-23 01-21-55 +0100). The issue I always faced was is that after I upgraded the servers it would run happily for the next 5-10 hours but for no reason FreeSWITCH would become unresponsive and stop accepting registrations. If you ran command like show channels, status etc the command would just freeze and there would be no response. Typically the freeze would occur when the servers were practically idle, looking at the logs you would see: [WARNING] sofia_reg.c:1524 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1050(a)domain.com] from ip x.x.x.x [WARNING] sofia.c:1639 MSG Thread 2 Started [WARNING] sofia.c:1639 MSG Thread 1 Started In attempt to resolve it I tried new hardware, changed from Centos to Debian, MySQL to Postgres, nothing I seemed to do fixed the issue. I started to run some network analysis on the server and what I noticed is about 20 minutes before the server freeze the traffic would spike from 100kbps to around 400kbps, on closer inspection I noticed that the 300kbps spike was actually between the server and a single IP on port 5060. Analysing the packets (tcpdump -s0 -A port 5060) I could see that one registration was sending SUBSCRIBE events at the rate of about 300-400 per second and FreeSWITCH was sending a NOTIFY back for each SUBSCRIBE, once the server froze the traffic would stop which is what made it hard to find as you had to be looking at the network traffic before the freeze. Once I knew the user causing the increased traffic flow I was able to work out that the user agent causing this was Zoiper (rev. 11137). It would appear Zoiper would get into a loop with the server and keep sending SUBSCRIBE packets for the message waiting indicator. I have spoken to Zoiper who have taken the issue on board and informed me it does not happen on Zoiper 3, but does happen on Zoiper 2, however they plan to fix it in the next release of Zoiper 2. They suggested for the moment to disable the MWI Subscribe option by doing "Go to the preferences -> select your sip account -> enable "show advanced options" andchange the mwi subscribe drop down to "disabled" The strange thing with all this is that the same Zoiper client never had a problem on version 1.0.head (git-e3698c2 2012-03-23 01-21-55 +0100) but as soon as you upgraded to 1.2 or even the current git versions you would find that FreeSWITCH and Zoiper would get into a SUBSCRIBE/NOTIFY Loop. I hope this information is of some help to someone else in the future and is able to ensure they do not have to go through the same nightmare I did. If anyone has any suggestions on ways to limit the SUBSCRIBE/NOTIFY responses so that you can avoid this happening in the future I would be very interested to hear, even if you have any other feedback that might have assisted in finding this issue quicker I would be very interested. Regards, Ashley Breeden URL Networks Pty Ltd WWW: http://www.url.net.au Twitter: @urlnetworks

3 4

TLS issue
by Vincent Xia 13 Sep '13

13 Sep '13

im trying to encrypt the voice call over FS in terms of SIP and RTP stream, what i do is simply follow instructions in http://wiki.freeswitch.org/wiki/SIP_TLS, the soft phone im using is PhonerLite 2.10, now the call establishment is OK, but after the called party answers, there's no voice at both sides, no RTP packets can be seen in wireshark which runs at FS box. The PhonerLite works fine with FS in uncrypted mode, so what's the problem with my configuration of SIP TLS?

2 2