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Seperate "new session" and "dial" in JS
by Francesco Facco de Lagarda 02 Jul '18

02 Jul '18

I am trying to make a call progess monitor I need to bind together the answering of incoming session The call to the outgoing session. What I am doing is basically trying to set a user defined variable "t3Id" to both sessions so I "marry then together" from the ESL At present I am doing (just a simplified extract of the code) var t3Id = "##randomly generated##"; session.answer(); session.setVariable("t3Id", t3Id); sessOut=new Session(##my dial string##) sessOut.setVariable("t3Id", t3Id); if(sessout.ready()) { bridge(session, sessOut); } session.hangup(); sessOut.hangup(); Unfortunately I noticed that IF I use sip services (I'm using Messagenet) the outgoing call never gets the variable set! Is it possible to do something like sessOut=new Session() sessOut.setVariable("t3Id", t3Id) sessOut.dial(##my dial string##) ??? I cant find any other parameter common to BOTH sessions to marry the two together!!!

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freeswitch internal profile hang freequently
by sukitha jayasinghe 02 Jul '18

02 Jul '18

Dear All, I have a freeswitch (1.6.20) server in production with db configured to postgresql through pgpool2. This system hangs time to time and has become a major issue. It happen only to the internal profile where our UACs connects through webrtc. After the incident profile stop responding to cli commands, only solution is restarting the entire application. I managed to get the back trace which indicate system waiting forever for profile database mutex. Further, It seems thread #19 is waiting for result from pgsql server cause this issue. Please find the back trace link below, https://pastebin.com/Uqe8HjZn Has anyone experienced this issue before, what is the solution for this. Best Regards, Sukitha.

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Failed Registration with status DNS Error [503]
by Paul Muaddib 29 Jun '18

29 Jun '18

Hi, how do I fix this problem: Failed Registration with status DNS Error [503]? I already search the net but couldn't find a proper answer. The only thing that I found was something about DNS SRV Resource Record but this was fixed in freeswitch a long time ago. *My Setup:* FreeSWITCH Version 1.6.19 Debian 7 wheezy Freeswitch(Debian) -> Router(OpenBSD) -> Internet *Failed Registration with status DNS Error [503]* 2018-06-29 19:27:21.732868 [NOTICE] sofia_reg.c:448 Registering telekom_voip nua.c:622 nua_register() nua: nua_register: entering nua_stack.c:529 nua_signal() nua(0x8c8c0c8): sent signal r_register nua_stack.c:569 nua_stack_signal() nua(0x8c8c0c8): recv signal r_register nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0xb5c1a298, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x8c8c0c8): adding register usage nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb6279478) nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, " sip-trunk.telekom.de.") called sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, " sip-trunk.telekom.de.") returned 1 entries nta.c:10398 outgoing_query_naptr() nta: for "sip-trunk.telekom.de" query " sip-trunk.telekom.de" NAPTR (cached) sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._ udp.sip-trunk.telekom.de.") called sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._ udp.sip-trunk.telekom.de.") returned 1 entries nta.c:10598 outgoing_query_srv() nta: for "sip-trunk.telekom.de" query "_sip._udp.sip-trunk.telekom.de" SRV (cached) sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, A, " sip-trunk.telekom.de.") called sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, A, " sip-trunk.telekom.de.") returned 1 entries nta.c:10803 outgoing_query_a() nta: for "sip-trunk.telekom.de" query " sip-trunk.telekom.de" A (cached) nta.c:1350 set_timeout() nta: timer set to 32000 ms nta.c:1348 set_timeout() nta: timer shortened to 5000 ms nua_stack.c:271 nua_stack_event() nua(0x8c8c0c8): event r_register 503 DNS Error nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x8c8c0c8): removing register usage nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb6279478) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2018-06-29 19:27:21.732868 [ERR] sofia_reg.c:2447 telekom_voip Failed Registration with status DNS Error [503]. failure #34 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2018-06-29 19:27:22.772957 [WARNING] sofia_reg.c:505 telekom_voip Failed Registration [503], setting retry to 30 seconds. nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (124810556) nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0xb60911e0) nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set *sofia_dig --tcp reg.sip-trunk.telekom.de <http://reg.sip-trunk.telekom.de>* Preference Weight Transport Port Address ================================================================================ 1 1.000 tcp 5060 217.0.26.165 2 1.000 tcp 5060 217.0.26.163 3 1.000 tcp 5060 217.0.26.133 *Gateway settings* <include> <gateway name="telekom_voip"> <param name="username" value=„XXXXXXXX“/> <param name="password" value=„XXXXXXXX“/> <param name="proxy" value="reg.sip-trunk.telekom.de"/> <param name="register-proxy" value="sip-trunk.telekom.de"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> </gateway> </include> Regards, Paul

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Unable to find modules.conf file after installation through RPM file
by Chandramouli P 29 Jun '18

29 Jun '18

Hello, I installed FreeSwitch through RPM file as below: sudo rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm sudo yum install freeswitch-config-vanilla sox freeswitch-sounds* freeswitch-lang* freeswitch-lua freeswitch-xml-cdr Everything is working fine. But, when I am trying to enable or disable the modules, I could not able to find the "modules.conf" files anywhere. Can anybody tell me how to enable/disable the Freeswitch modules, if we install the FreeSwitch through RPM? Thanks in advance. Regards, CM. <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Re: [Freeswitch-users] Enterprise/Production Quality?
by David Villasmil 28 Jun '18

28 Jun '18

Hello all, Is this true??? Bug fixes aren't available for the general open source community for EIGHTEEN MONTHS??? This seems REALLY wrong to me... I'm pretty sure there are better ways of making money from freeSWITCH... Now I'm very concerned, and probably a lot of people will, if/when they find out about this... David On Tue, Jun 26, 2018, 02:57 Matteo via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Matteo <mbrancaleoni(a)voismart.it> > To: freeswitch-users <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 20 Jun 2018 11:50:26 +0200 (CEST) > Subject: Re: [Freeswitch-users] Enterprise/Production Quality? > Well I agree too. > > Since the introduction of the FSA (term not cited anywhere on website), we > started evaluating a subscription. > > But a lot of things are really, really vague and keeping us away: > > * What is in the FSA? What we get for the money? > * What about prices? > * What is the release cycle? When things gets pushed into the OSS version? > * What is the FSA? please add a blog post to clarify this shift in policy > and how things "works" from now on. > > Basically there's no communication about that and everything appears > really vague. > > Other OSS projects with similar policy have a very clear release path, > what is in, what is not, > when new things will appear in OSS and so on. > > And then a suggestion: > Why not offer a "basic" FSA with just access to latest git? > And not provide dedicated support, but use traditional channels? > > This should be cheaper and interesting for companies that are autonomous > into investigating and fixing issues by themselves. > > And about bugfixes... well I agree with others... they should be released > because affects something that's already out. > > Not doing that may backfire because the OSS product may be perceive buggy > (especially if using latest features which may not be debugged properly yet) > > Keep in mind that from a company perspective, if the things are free you > can be loose on how the project is handled, because is free after all. > But if you want to get payed (which is completely natural and needed to > keep things go on), you must be very clear and communicative on what you > give for the price. > > Just my 2 cents. > > Matteo > > ----- Il 30-mag-18, alle 21:00, Michael Avers michael(a)mailworks.org ha > scritto: > > > I agree. I think they are doing it wrong. I obviously believe they need > to make > > money and get paid for their work, I have no problem with that at all, > and my > > company has bought quite a few licenses of their commercial modules and > we > > attend Cluecon every other year. So we have no problem paying but we > need to > > know what we are paying FOR. > > > > This kind of vague state of things where you just don't know if bugs are > now > > going to be fixed for everyone or just for the privileged ones is not a > good > > path to go down. > > > > Bug fixes really should be made available to everyone (assuming they are > in > > previously public modules, of course). > > > > The Freeswitch team should focus their efforts on creating commercial > modules, > > ready-made apps and setups, pro versions of older modules, say enhanced > > mod_callcenter or whatever. Things like that. But to tell someone to get > a > > premium subscription just to get a bug fixed... that's simply wrong. > > > > Just my 2 cents > > > > Mike > > > > > > > > > > On Wed, May 30, 2018, at 11:37 AM, William Simon wrote: > > > > > > > > I thought stability fixes were going to be included in the open source > release > > whereas new features are covered under FSA release cycle. What I am > seeing here > > is unfortunate. Open source users should just accept instability for the > > 18-month release cycle? > > > > > > > > > > > > On May 30, 2018, at 12:35 PM, Michael Jerris < mike(a)jerris.com > wrote: > > > > Michael- > > > > This is a specific bug that I know we have fixed. We spent months of work > > tracking it down, I am very familiar with the issue. This issue is not > at all > > with verto, and is specifically with the sip secure web socket support. > We have > > never recommended the use of sip web socket support for webrtc, we think > that > > verto is typically a better solution, and is more stable. > > > > Mike > > > > > > > > > > > > On May 30, 2018, at 12:16 PM, Michael Avers < michael(a)mailworks.org > > wrote: > > > > LOL... The issue he describes can be due to several different reasons. I > don't > > even think we have enough information at this point to determine it let > alone > > narrow it down to one specific bug fix. > > > > Is saying "Oh this is fixed in our paid product" going to be the > standard moving > > forward? > > > > In any case, a response such as this basically tells potential Verto > users that > > they shouldn't bother because it's going to break at 150 concurrent users > > anyway. > > > > Mike > > > > > > On Wed, May 30, 2018, at 8:58 AM, Michael Jerris wrote: > > > > > > > > Geoff- > > > > I believe this issue is fixed in > https://freeswitch.org/jira/browse/FS-10762 > > which is available to FSA customers. > > > > Mike > > > > > > > > > > On May 30, 2018, at 10:45 AM, Geoff Mina < gmina(a)connectfirst.com > > wrote: > > > > Is anyone out there actually using FS successfully in an enterprise > environment? > > > > We have deployed a handful of servers in an extremely simple > configuration to > > allow standard SIP infrastructure to communicate with WebRTC clients. > > > > We run ~150 concurrent users per host and we can’t go a week without > something > > in the core of Sofia failing. We have seen hung profiles that simply > don’t > > respond to REGISTER requests (tried every suggested tweak to no avail) > as well > > as seeing FS hang every call in a RINGING state without actually ringing > the > > end client. > > > > These seem like pretty fundamental components we are struggling with. > Our use > > case seems quite simple - yet the software (1.6.19) seems like it has > never > > even seen a production deployment. > > > > Anyone out there have a drastically different experience? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.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 > > > > > > > > “The information transmitted is intended only for the person or entity > to which > > it is addressed and may contain proprietary, business-confidential and/or > > privileged material. If you are not the intended recipient of this > message you > > are hereby notified that any use, review, retransmission, dissemination, > > distribution, reproduction or any action taken in reliance upon this > message is > > prohibited. If you received this in error, please contact the sender and > delete > > the material from any computer.” > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.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 Services > > sales(a)freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://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 > > https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Matteo via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 25 Jun 2018 17:57:02 -0700 (PDT) > Subject: Re: [Freeswitch-users] Enterprise/Production Quality? > _________________________________________________________________________ > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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Silent calls, non-silent recordings
by David P 27 Jun '18

27 Jun '18

We've started seeing calls that aren't audible but in which both legs of the video conference *are* audible in recordings made at the same time. Here are some of the variables: 1. We have two clouds with one FS in each 2. Both clouds have been configured for TLSv1.2 + SRTP 3. The network public/private IPs and port openings are the same 4. Bria for Windows works fine with "verify TLS certificate" on 5. Bria for Mac gets no audio regardless of security setting 6. A verto page pointed at one of the clouds works for Chrome/Windows and Safari/MacOS but doesn't work for either when pointed at the other cloud 7. I have looked at the FS log at debug level for one of these silent calls, and all of the codec negotiations go as expected and there are no errors. The problem I'd most like help with is finding possible reasons why webRTC calls are silent but have non-silent recordings.

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Voicemail mwi multiple mailboxes notify one user
by Mircea BH 26 Jun '18

26 Jun '18

Hello, I would like to know if there is any possibility to notify one user when messages are left in multiple mailboxes? I want to have just one registered user which should monitor multiple voicemail mailboxes. Thank you for your time. Best regards, Mircea

1 0

h264 video call between endpoints with one end at bad resolution of 352x288
by antonio 26 Jun '18

26 Jun '18

Hi, Calling between two Grandstream GXV3275 i expected to have a video call at 1280x720, but the video that is present at destination as the resolution of 352x288 and the image as poor quality, originator is sending video at 1280×720. If i swap the devices the result is the same, the destination always get a bad resolution of 352x288 but the originator receives the video at expected resolution of 1280x720. If i set rtp_direct between the two endpoints it work as expected. I also tested a conference room and force the it resolution to 1280*x720 it work well, if i don't force the canvas-size i see: 2018-06-26 13:55:02.077824 [WARNING] mod_conference.c:3157 Unspecified video-canvas-size, falling back to 1280x720 And the conference is done at 1280x720. I've open a jira where you can find a full log with this behavior: https://freeswitch.org/jira/browse/FS-11140 I don't understand why there is transcoding and mod_av is used, the codec negotiated is the same.. i understand that the resolution is at 352x288 for the first frames sent by the phone but then it send them at 1280x720, mod_av should detected and adjust the new resolution, no? Anyone experience this behavior? -- Saludos / Regards / Cumprimentos Anónio Silva

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Problems with NAT
by Leandro Campos 26 Jun '18

26 Jun '18

Recently we are testing the Freeswitch with the Astpp for change us servers (we use asterisk with a2billing). In the migration process, we check some customers don’t register or don’t receive inbound calls. In a2billing works fine. I think is something about NAT. Someone help me? Atenciosamente, Leandro Campos CEO (11) 4118-6267 https://www.nvoip.com.br

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Peer Subject Mismatch (incoming connection)
by Xenia Obolenskaya 23 Jun '18

23 Jun '18

Hi All, It is strange, very strange history. When I made the blank settings of value, type of <param name="tls-verify-in-subjects" value=""/> the check of peer name in the manner <param name="tls-verify-policy" value="subjects_all"/> works good. The same favorable outcome I can observe if value settins are of kind <param name="tls-verify-in-subjects" value="sip.obolenskaya.su"/> that is to say pattern-matching filtering. Now Peer Subject in incoming connection matches! But I nowhere could find the similar clarifications. Thank All! Regards, Xenia Obolenskaya

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